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The final coverage and associated performance of a single frequency networks (SFNs) is a joint result of the properties of all transmitters in the SFN. Due to the large number of parameters involved in the process, finding the right configuration is quite complex. The purpose of this paper is to find optimal SFN network configurations for second generation digital terrestrial broadcast system (DVB-T2). Offering more options of system parameters than its predecessor DVB-T, DVB-T2 allows large SFN networks. However, self-interference in SFNs gives rise to restrictions on the maximum intertransmitter distance and the network size. In order to make optimum use of the spectrum, the same frequency can be reused over different geographical areas—beyond the reuse distance to avoid co-channel interference. In this paper, a methodology based on theoretical network models is proposed. A number of network architectures and network reference models are considered here for different reception modes in order to study the effects of key planning factors on the maximum SFN size and minimum reuse distance. The results show that maximum bitrate, network size, and reuse distance are closely related. In addition, it has been found that the guard interval is not the only limiting parameter and that its impact strongly depends on the rest of DVB-T2 mode parameters as well as on the network characteristics (equivalent radiated power, effective height, and intertransmitter distance). Assuming that the carrier to noise ratio requirements are in the vicinity of 20 dB and bitrates over 30 Mb/s, it has been found that the network can be as large as 360 \times 360 km (delivering 39.2 Mb/s) or even 720 \times 720 km (delivering 37.5 Mb/s). The reuse distance will also have a complex dependency on the DVB-T2 mode and especially the network parameters, ranging from below 100 to 300 km. [ABSTRACT FROM PUBLISHER]

Multimedia Broadcast Multicast Service (MBMS)-dedicated cell is a Long Term Evolution (LTE) based broadcast transmission mode with both high transmission efficiency and good compatibility with LTE unicast systems. However, since the physical design of Cell Acquisition Subframe (CAS), which is the control part of an MBMS-dedicated cell transmission period, does not take into account the long delay spread channels in Medium Tower Medium Power (MTMP) and High Tower High Power (HTHP) cells, receivers would probably suffer from multipath aliasing errors in practical transmission. This paper analyzes the impact of aliasing errors on channel estimation (CHE) accuracy and proposes an Aliasing-Elimination CHE algorithm for CAS reception. In the proposed algorithm, the pre-decoded Physical Broadcast Channel (PBCH) is reconstructed and utilized to generate a complete multipath distribution within the entire OFDM symbol length. By using this full-symbol length multipath distribution, the proposed algorithm eliminates aliasing errors effectively. Simulation results and complexity analysis show that the proposed CHE algorithm achieves a significant performance improvement compared to conventional CHE algorithms in long delay spread channels, with acceptable increment in complexity. [ABSTRACT FROM AUTHOR]

Most wireless microphones operate on vacant television broadcasting channels in the very high frequency and ultrahigh frequency bands, as secondary users operating on the no-interference and no-protection basis. Some wireless microphones share the 2.4 GHz band with other wireless devices. The 698–806 MHz band was repurposed when television broadcasting switched from analog to digital (and the 600 MHz band may be repurposed in the near future), resulting in less spectrum available for wireless microphone operations. To keep wireless microphones operating effective in this environment, engineers, researchers, and regulators require comprehensive knowledge of the spectrum, interference, and regulations regarding wireless microphones. This paper first presents the characteristics of the frequency bands currently allocated for wireless microphone operations. Next, interference that may be experienced by wireless microphones and frequency coordination required to allow multiple microphones to work together at the same site without interference are presented. Finally, it summarizes the regulations in Canada, United States of America, European Union, and Australia for wireless microphones regarding allowable transmitter frequency, power, bandwidth, channel masks, maximum frequency deviation for FM, and spurious emissions. [ABSTRACT FROM PUBLISHER]

Zapping-delay is an important factor contributing to the quality-of-experience (QoE) of any multimedia service; the longer the zapping-delay, the worse is the QoE. Therefore, for a good QoE, it is necessary to bind zapping-delay within acceptable bounds. Coded video uses a motion compensated prediction structure that contributes to the zapping-delay. In DVB-H, zapping-delay is further increased because of the time-sliced burst transmission architecture. Hence, any attempt to bind zapping-delay must consider both the prediction structure of coded video, as well as the time-sliced transmission architecture of DVB-H. This paper analyzes prediction structure of video as binary relations generating a prediction graph. Reachability concepts of graph theory are then applied to construct binary valued indicator functions. These indicator functions inform about the reception, decoding, and play-out states of each access-unit in the video sequences. zapping-delay is studied by dissecting the components that it is composed off. Conditions for minimizing each of them individually, while the final zapping-delay does not exceed a known bound, is found. Emphasis is also laid on gradual quality enhancement after the display of the new program has started. [ABSTRACT FROM AUTHOR]

This paper investigates the impact of different audio codecs typically deployed in current digital audio broadcasting (DAB) systems and web-casting applications, which represent a main source of quality impairment in these systems and applications, on the quality perceived by the end user. Both subjective and objective assessments are used. Two different audio quality prediction models, namely Perceptual Evaluation of Audio Quality (PEAQ) and Perceptual Objective Listening Quality Assessment (POLQA) Music, are evaluated by comparing the predictions with subjectively obtained grades. The results show that the degradations introduced by the typical lossy audio codecs deployed in current DAB systems and web-casting applications operating at the lowest bit rate typically used in these distribution systems and applications seriously impact the subjective audio quality perceived by the end user. Furthermore, it is shown that a retrained POLQA Music provides the best overall correlations between predicted objective measurements and subjective scores allowing to predict the final perceived quality with good accuracy when scores are averaged over a small set of musical fragments ( \mathbf R = 0.95 ). [ABSTRACT FROM PUBLISHER]

To cope with increasing demands for spectral efficiency, multiple-input multiple-output (MIMO) technology is being considered for next generation terrestrial broadcasting television systems. In this paper, we propose a MIMO channel-precoder that utilizes channel statistical structure and is suitable for terrestrial broadcasting systems, while being potentially transparent to the receivers. The performance of the channel-precoder is evaluated in a wide set of channel scenarios and mismatched channel conditions, a typical situation in the broadcast setup. Capacity results show performance improvements in the case of strong line-of-sight scenarios with correlated antenna components and resilience against mismatched condition. Finally, we present bit-error-rate simulation results for state-of-the-art digital terrestrial broadcast systems based on digital video broadcasting—next generation handheld to compare the performance of single-input single-output, 2 \times 2 and 4 \times 2 MIMO systems and proposed MIMO channel-precoder. [ABSTRACT FROM PUBLISHER]

This paper investigates the effects of crest factor reduction and digital predistortion on the performances of communication and broadcasting systems. The overdrive of digitally predistorted power amplifiers is proposed for power efficient linear amplification of amplitude modulated signals having high peak to average power ratio. This is achieved by adding, at the input of the digital predistortion system, a clipping function to allow for an overdrive of the linearized power amplifier while meeting spectrum emission mask linearity requirements. The proposed technique is experimentally validated using a GaN Doherty power amplifier driven by long-term evolution (LTE) signals. Measurement results show that the overdriven power amplifier can deliver up to 2.5 dB extra output power, up to 80% enhancement in the power utilization factor, and up to 20% relative increase in the power efficiency while being compliant with the linearity requirements of the LTE standard. [ABSTRACT FROM PUBLISHER]

In wireless push-based systems traffic is asymmetric meaning that the server has much more bandwidth available, as compared to the clients. Since it is often not possible or desirable for the clients to send any requests, the server should broadcast the data periodically. The performance of this system relies heavily on the proper scheduling of the broadcast data. The well known algorithm proposed by Vaidya and Hameed battles this challenge and achieves state-of-the-art performance. However, this performance comes with a computational complexity linear to the number of data items that are broadcast. Furthermore, Vaidya’s and Hameed’s proposed solution for this time complexity degenerates performance. This paper proposes a scheduling algorithm for wireless push-based data broadcasting systems, which performs near optimal and has low computational complexity. Performance evaluation results are presented to demonstrate its efficient performance despite its low computational demands. [ABSTRACT FROM PUBLISHER]

As different types of wireless networks are converging into an all-IP network, i.e., the Internet, it can be expected that in the near future video-on-demand (VoD) will be widely applied to many interesting services, and users can access these services using heterogeneous terminals via heterogeneous wired/wireless access networks. Many periodic broadcasting protocols have been proposed to reduce the implementation cost of VoD systems. However, most of the protocols assumed homogeneity for user terminals, while in practice, user terminals are usually quite different in their processing power, buffer space, and power. To address this problem, a few periodic broadcasting protocols providing the same video quality for all heterogeneous clients have been proposed recently. In this paper, we proposed a novel heterogeneous VoD broadcasting technique called Catch and Rest (CAR) to accommodate bandwidth heterogeneity without sacrificing user video quality. Then, we provide mathematic analysis to calculate the client bandwidth and buffer space requirements of CAR. Finally, we present our performance evaluation results for CAR. Our results show that under the same system resources (i.e., server and network bandwidth), CAR provides more uniform and acceptable service latency for all heterogeneous clients compared to previous works. [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]

Nowadays more and more TV terminal devices for digital terrestrial broadcast are also equipped with broadband network interfaces. In view of this fact, we developed a novel redundancy on demand (RoD) system to improve the reception of terrestrial broadcast signals with the help of additional redundancy data transmitted via broadband networks. We propose various forms of redundancy, which can be provided on request by the receiving terminal in case of reception errors. RoD does not require any changes of the current broadcast network structure and is compatible with conventional receivers. Simulation results show that a small portion of additional redundancy data provided by the broadband network leads to significant performance improvements of the reception, making high data rate broadcasting feasible in a realistic environment. RoD introduces a new degree of freedom for network planning helping to reduce transmission cost and network energy consumption. [ABSTRACT FROM PUBLISHER]

A method to design channel interleavers based on span properties and mutual information is presented. Resulting interleavers enjoy better burst error control and can mitigate the effect of regular error patterns. An application example based on the 2nd generation digital video broadcasting terrestrial (DVB-T2) standard is elaborated. Four channel interleaving structures are generated, resulting from a joint optimization of the interleaver span properties in the time and frequency domains, while guaranteeing good interaction with the bit interleaver. Proposed solutions are compared to the standardized DVB-T2 channel interleaver in terms of performance, latency, and complexity. A significant improvement can be observed in severe channel conditions, especially over time and frequency-selective channels with erasures. Moreover, the complexity and latency of the proposed channel interleavers are reduced compared to the original DVB-T2. [ABSTRACT FROM AUTHOR]

We present the latest developments in the physical layer evolution for the transmission of Multimedia Broadcast Multicast Service (MBMS) over fifth generation (5G) cellular networks. We first provide an overview of how the MBMS physical layer has evolved during the initial releases of the Third Generation Partnership Project (3GPP) specifications and provide motivations and insights for key developments that have helped mature MBMS into a competitive broadcasting solution. We provide an overview of dedicated MBMS carriers for Multimedia Broadcast multicast service Single Frequency Network (MBSFN) transmission as well as Orthogonal Frequency Division Multiplexing (OFDM) numerologies with longer cyclic prefixes to support MBSFN transmission over large geographical areas with large distances between cooperating cell sites. Next, we describe the most recent enhancements to the MBMS physical layer, addressing advanced use cases in the recently completed release of the 3GPP specifications (Release 16). First, we describe the physical layer design for enabling MBSFN-based MBMS services for rooftop receivers with inter-site distances between cooperating cell sites as large as 125 km. In this realm, we demonstrate the need for an OFDM numerology with a 300-microsecond cyclic prefix to communicate effectively over channels with very large delay spreads. Next, we explain how support for high mobility up to 250 kmph was added to the standards by using an OFDM numerology that gracefully trades off inter-symbol interference due to large delay spreads and inter-carrier interference due to large doppler spreads to achieve better performance than existing numerologies at high speeds. Finally, we describe the enhancements that were made to the Cell Acquisition Subframe (CAS)—specifically, the introduction of larger aggregation levels for the Physical Downlink Control Channel (PDCCH) and the support of Physical Broadcast Channel (PBCH) repetitions—that significantly increase the coverage for the control and system information associated with MBMS that are carried by these subframes. [ABSTRACT FROM AUTHOR]

These instructions give guidelines for preparing papers for this publication. Presents information for authors publishing in this journal. [ABSTRACT FROM PUBLISHER]

We study channel pre-inversion and vector perturbation (VP) schemes for large-scale broadcast channels, wherein a transmitter has M transmit antennas and is transmitting to K single-antenna non-cooperating receivers. We provide results which predict the capacity of MMSE pre-inversion as \boldsymbol K\rightarrow \infty . We construct a new VP strategy, max-SINR VP (MSVP), which maximizes a sharp estimate of the signal-to-interference-plus-noise ratio. We provide results which predict the performance of MSVP and demonstrate that MSVP outperforms other VP methods. Lastly, we combine MSVP with the low-complexity sorted QR precoding method to show that MSVP has the potential to efficiently deliver data to a very large number of users at close to channel capacity. [ABSTRACT FROM PUBLISHER]

The impact of channel estimation errors is investigated in two-way relay networks (TWRNs) with two different channel state information (CSI) estimation schemes, namely separate and cascaded. In the former scheme, the relay node is chosen in accordance with the CSI of the individual channels between the source nodes and the relay node. By contrast, in the latter scheme, relay node selection is performed in accordance with the CSI of the cascaded channel between the two source nodes. For both estimation schemes, the effect of the CSI estimation error on the outage probability is examined by means of the Cramér–Rao lower bound. Moreover, in contrast to previous studies, the source nodes in the TWRN are assumed to have different estimation errors. The numerical results confirm the validity of the derived expressions and show that the presence of different estimation errors at the two source nodes has a significant effect on the performance of the TWRN. [ABSTRACT FROM PUBLISHER]

Watching television on a portable device is possible on broadcast and cellular systems. With broadcast technologies, due to the absence of a feedback channel it is not possible to guarantee an error free reception at the receivers. With cellular technologies, the transmission is based on unicast and the same content is transmitted as many times as the number of people who are watching the same program in the cell. We propose to combine both technologies and consider the concept of repairing a broadcast data flow for dual-mode (cellular and broadcast) smartphones in real time in order to ensure a better quality of service. We propose to use the constrained application protocol (CoAP) and specify a real-time flow repair (RFR) service based on CoAP in the cellular networks. We describe the architecture to provide such a service. We analyze the load generated in the cellular radio access network due to the retransmission of the packets lost on the broadcast system and also develop a model for the evaluation of the residual errors in the RFR operations. [ABSTRACT FROM PUBLISHER]

We investigate the design of a broadcast system in order to maximize throughput. This task is usually challenging due to channel variability. Forty years ago, Cover introduced and compared two schemes: time sharing and superposition coding. Even if the second scheme was proved to be optimal for some channels, modern satellite communications systems such as DVB-SH and DVB-S2 rely mainly on a time sharing strategy to optimize the throughput. They consider hierarchical modulation, a practical implementation of superposition coding, but only for unequal error protection or backward compatibility purposes. In this article, we propose to combine time sharing and hierarchical modulation together and show how this scheme can improve the performance in terms of available rate. We introduce a hierarchical 16-APSK to boost the performance of the DVB-S2 standard. We also evaluate various strategies to group the receivers in pairs when using hierarchical modulation. Finally, we show in a realistic case, based on DVB-S2, that the combined scheme can provide throughput gains greater than 10% compared to the best time sharing strategy. [ABSTRACT FROM PUBLISHER]