ABSTRACT. In this paper, we present an efficient transmission method of multiple broadcasting services using Layered Division Multiplexing (LDM) and Scalable High Efficiency Video Coding (SHVC) for the next generation terrestrial digital broadcasting such as Advanced Television Systems Committee (ATSC) 3.0. Based on a two-layer LDM technology, each Physical Layer Pipe (PLP) carries each of Base Layer and Enhancement Layer video streams of SHVC, in order to maximize channel utilization when multiple broadcasting services are delivered in a single radio frequency (RF) channel. A performance analysis is conducted when the proposed combination of LDM and SHVC is compared with other combinations such as Time Division Multiplexing (TDM) and/or HEVC simulcast broadcasting. A constant quality measurement is introduced to evaluate the performance of video codec in conjunction with a physical layer system. The proposed combination of LDM and SHVC is implemented in hardware and verified in a real field environment.

ABSTRACT. As one of the candidates of the next generation terrestrial UHDTV broadcasting system in Japan, the FDM(Frequency Division Multiplexing) system that is an extension of the current standard for HDTV broadcasting(ISDB-T: Integrated Service Digital Television Broadcasting – Terrestrial) is being considered. In this system, the 6MHz transmission band is divided into 36 small bands that is called ‘segment’ and the transmission is performed using 35 or 33 segments. By combining multiple segments flexible band allocation to multiple contents is made possible and currently multiplexing of maximum 3 streams is assumed. Typical use of the segments is that the contents for stationary reception and mobile reception are allocated using the segments, and the system that the central 9 segments are used for mobile reception and the other 26 segments are used for stationary reception similar to the current ‘ISDB-T one segment service’ is currently discussed. By partially receiving central 9 segments, simple implementation of mobile receiver becomes possible. In this paper, the scheme that performs LDM(Layered Division Multiplexing) in the central 9 segments is proposed in order to increase the flexibility of transmission and to increase the performance of transmission under specific conditions. As an initial evaluation of the proposed scheme, the performance of the proposed scheme was evaluated under Gaussian channel and improvements of performance under specific conditions were confirmed.

ABSTRACT. ATSC 3.0 includes Layered Division Multiplexing (LDM) in addition to the traditional Time Division Multiplexing (TDM) technique in order to offer simultaneous stationary and mobile services. LDM has been extensively proved to improve TDM even when LLR PDF formula is not optimized for multilayer signals. The LDM optimized PDF formula improves the performance in return for higher implementation requirements. The main objective of this paper is to analyze the performance of using semi-optimal LLR PDF formulas for LDM signals in order to find the best trade-off between performance and implementation requirements. In addition, the new gain in comparison to TDM is also presented.

ABSTRACT. In this paper, we propose a new method in which the recently proposed non-orthogonal multiple access (NOMA) is combined with the conventional Alamouti scheme. The developed analytical results and simulation both show that if the targeted data rates and the allocated power satisfy an inequality constraint, then our proposed method can effectively increase the diversity order of NOMA from 1 to 2, the same as the Alamouti scheme used in conventional orthogonal multiple access(OMA). Moreover, we define the .-achievable rate region as criterion to evaluate the rate gain of the proposed method, and derive closed-form expressions of the .-achievable rate region of both the proposed method and OMA-based Alamouti scheme. Simulation shows that the .- achievable rate region of OMA-based Alamouti scheme is strictly a subregion included in that of the proposed method. The conclusion of this paper is that NOMA can be effectively combined with Alamouti scheme to get the benefits of both the capacity gain of NOMA and the diversity gain of Alamouti scheme.

ABSTRACT. In this paper we describe a novel approach to create standard-compliant H.264/MVC video streams in real-time by distributing most of the computational complexity amongst multiple H.264/AVC encoders which can be part of the capturing cameras or separate devices inbetween the cameras and the transcoder. By exploiting the structural similarities between H.264/AVC and H.264/MVC the transcoder only has to decode its individual input streams at a very high level. With minimal changes to the input streams a standard-compliant multi-view video stream is created. This method is efficient enough that a high number of input streams can be transcoded on a consumer multi-core PC.

ABSTRACT. Current video services are still controlled in an old-fashioned way using keyboard and mouse for computers and remote control for TV sets. This paper presents more attractive and intuitive interaction methods using commercially available motion sensing input devices. These devices are typically based on a webcam-style add-on peripheral thereby enabling facial recognition, gesture control, and speech recognition. These technologies are applied in this research to automatically authenticate a user, enable video control (play, pause, seeking), and browsing, selecting, and rating content by hand gestures or voice commands. By monitoring the user's gaze and using emotion recognition techniques, the user's interests and engagement with the content can be estimated. This is interpreted as implicit user feedback for the video content, and establishes an automatic feedback channel, which can be used for content personalization and recommendation. User tests showed an accurate recognition of voice and gestures and confirmed the attractiveness and intuitiveness of these techniques for end-users.

ABSTRACT. Light fields are gaining more impact in today’s filming technology due to post-processing possibilities such as software refocus, perspective shift, reconstruct 3D images, and depth scaling. Light fields are captured light rays at every point in space travelling in every direction, therefore a large amount of data is recorded for representation. For exchange, broadcast and storing of such imagery data, we lack efficient compression techniques. This paper proposes a light field compression approach based on low-complexity preprocessing by pseudo-temporal sequencing frames suitable for state-of-the-art codecs. An efficient novel approach of circular reordering is introduced where the redundancy between frames is maximized and exploited by low-delay HEVC predictive coding. Experimental results showcase a consistent increase in both PSNR and SSIM for several light fields and outperforms available schemes with pseudo-temporal sequencing using HEVC. An increase upto 1 dB gain compared to state-of-the-art reordering techniques using HEVC is achieved.

ABSTRACT. The extremely growth of mobile streaming data over the world imposes a significant burden on power consumption of smart mobile devices. In order to obtain the more precise energy consumption model on the smart devices and analyze the energy consumption of different quality level mobile streams in the different wireless environment (e.g. WiFi and LTE), an open-source power consumption measurement platform is developed for the real-time multimedia content delivery and smartphone power consumption measurement, and the evaluation testing of the proposed platform in Dublin City University Campus are also performed.

ABSTRACT. This paper proposes a distributed spectrum sensing method based on an IoT architecture. The proposed architecture comes in support for short-range video broadcasting for delivering commercial contents to several TV devices in an indoor scenario, involving all sensing-enabled objects in a distributed sensing procedure. The main contribution of this paper is the creation of new templates for the virtual objects (VOs) in order to represent the spectrum sensing capabilities of devices such as Smart TVs and Raspberry Pi computational platforms and the creation of mashed-up VOs (micro-engines - MEs), able to find the relevant VOs and to decide whether the sensed channels can be used or not for secondary communications. Finally, a distributed set-up based on Raspberry Pi computational platforms has been developed to test the proposed architecture.

ABSTRACT. The DRX (Discontinuous Reception) mechanism is the currently used technique for power saving in mobile devices. This approach is also utilized for LTE-Advanced and efforts are being made in order to enhance the performance of this mechanism. The currently used algorithm saves power in trade-off with delays. Using the same DRX configurations all the time with various traffic characteristics is not efficient when the aim is to achieve better performance. Lower delays and lower power consumption levels could be realized by implementing an efficient adaptive DRX approach. The proposed scheme for LTE-Advanced implements adaptive DRX configuration switching based on the traffic characteristics by tuning the DRX configurations to achieve the required power saving while at the same time keeping delays to a minimum and improving the mobile device’s performance. Preliminary results indicate that a multi-DRX configuration is an effective way to achieve the best performance. In this contribution, the performance of the power saving mechanism for different types of video streaming such as: low resolution, high resolution and Full HD quality videos will be presented. The DRX configurations are switched based on the traffic characteristics in order to avoid limitations, such as overhead and the trade-off between delays and power saving, while taking into account the performance of other running processes such as HARQ, scheduling and handovers.

ABSTRACT. Lately Video Sensor Networks (VSN) are increas- ingly being used in the context of smart cities, smart homes, for environment monitoring, surveillance, etc. In such systems, due to the limitations related to video sensor node resources, the most important factor is energy efficiency. Sensing, information processing and data transmission are the main contributors to energy consumption in a video sensor node. Among these, wireless transmissions play a dominant role. An effective way of reducing energy consumption is to adjust the active time duration of node’s radio transmission. However, the main drawback of this method is that it affects streaming quality in terms of throughput and delay. Therefore, one of the most important challenges when designing an energy-aware VSN is the trade-off between energy consumption and video delivery quality.

This paper proposes an Enhanced scheme for Adaptive Multimedia Delivery (eAMD) that dynamically adjusts the wake-up/sleep duration of video sensor nodes based on the node remaining battery levels and network traffic conditions. A Markov Decision Process (MDP)-based framework is used to formulate the problem and an innovative algorithm based on Q-Learning is proposed to find the optimal policy for video sensor nodes. Using both a systematic and algorithmic approach, our proposed system architecture and algorithms hold the potential to improve the trade-off between video streaming quality and energy efficiency in comparison with other state-of-the-art adaptive video based algorithms.

ABSTRACT. This paper is on the calculation of exact bit error rate (BER) of TCM M-ary APSK applicable to the DVB for multimedia data transmission. The current DVB-S2/S2x standards utilize the concatenated FEC coding (LDPC + BCH) for QPSK, 8PSK, M-APSK (M is 8 to 256). However this complicated coded modulations can be effectively combined into a single process using TCM. Although standards specify the average BERs using simulation, currently no closed forms on the exact BER calculation can be found. Our approach is unique since practical pairwise error probability-based product trellis algorithm is used to yield the exact BERs of 2 and 4 state TCM-MAPSKs (M=4, 8, 16, 32, 64) with a proposed set partitioning under the constraint of identical total symbol energy regardless of M . The calculated results confirm the accuracy/effectiveness of the proposed method.

ABSTRACT. 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 broadcasting systems. High throughput and low latency are two basic requirements that the future systems need to meet. 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. The latency of parallel turbo decoder after implementation can be as less as 23.2us at a clock rate of 250 MHz and the throughput can reach up to 6.92Gbps.

ABSTRACT. Low-density parity check (LDPC) codes, especially quasi-cyclic (QC-) LDPC codes, have captured lots of attentions for their capacity-approaching capability and low decoding complexity. However, conventional QC-LDPC based systems optimized for independent demapping schemes always show poor performance with iterative demapping. In this paper, spatially coupled (SC-) QC-LDPC code, which has provably capacity-achieving capability and universality, is proposed to be the channel code, in order to achieve a good tradeoff between multiple-input multiple-output (MIMO) bit-interleaved coded modulation (BICM) and BICM with iterative demapping (BICM-ID) schemes. The SC-QC-LDPC code is designed and optimized with the aid of multi-edge type density evolution, which can provide more accurate performance analysis than conventional single-edge type density evolution. Through simulations, the gaps between signal-to-noise (SNR) thresholds and the corresponding constellation constrained capacities with different spectral efficiencies are obtained. Given the spectral efficiency, the gap of MIMO BICM scheme is only 0.01~0.06 dB away from that of MIMO BICM-ID scheme, which indicates a good tradeoff between MIMO BICM and BICM-ID schemes is achieved with the proposed SC-QC-LDPC code.

ABSTRACT. Hybrid broadcast broadband TV (HbbTV) is a technique providing Push-VOD services over an interactive hybrid TV. These services are broadcast using File Delivery Protocol (FDP) characterized by three levels of data representation and give rise to three loss distribution that may result in QoS degradation and poor Forward Error Correction (FEC) recovery capabilities. In this paper, we first analyze the behaviour of FDP protocol system over a large variety of bursty erasure channels to characterize and to understand effectively the loss patterns seen within FDP protocol and at the AL-FEC input in order to be able to dimension the AL-FEC codes. Then, we propose and investigate a novel solution based on RS AL-FEC codes for FDP protocol to increase the robustness of HbbTV based Push-VoD services delivery over DVB networks. This paper represents a first reference work for a reliable HbbTV based Push-VOD services delivery over burst-loss channels.

ABSTRACT. While virtual reality (VR) videos have attracted significant amount of attention and efforts, the development of VR is still quite challenging. Among the challenges, the need for enormous bandwidth in live streaming hasn’t yet been addressed satisfactorily. To tackle this aspect, we propose a 360 VR live streaming system based on MPEG media transport (MMT) to guarantee the delivery of VR video as well as the user experience. Within the proposed scheme, we divide the integrated VR video into multiple partitioned videos, and encode and encapsulate the partitioned videos into different MMT Assets. An MMT signaling message is introduced to indicate the spatial relationship among the partitioned Assets. The server selectively delivers the partitioned Assets that cover the client’s current viewport with high bitrate. Meanwhile, other Assets are also delivered with low bitrate to the client for covering the entire view. We compare the bandwidth used by the developed system in delivery and PSNR of the delivered Assets with benchmark approaches in VR streaming.

ABSTRACT. In addition to the start of the next-generation broadcast service and the generalization of the video delivery service using the broadband, general users are sharing many private video content. We consider the system that utilizes effectively the increasing video content. Regardless of whether the sender is a broadcast company, a delivery company, or a general users, we assume that an innumerable amount of video content with metadata of the absolute time exist in the system. We describe the functional requirements to realize this system and the stored formats which is one of the requirements. We implemented the proposed format and performed basic verification.

ABSTRACT. Nowadays, camcorder piracy has great impact on the motion picture industry. Although some watermarking technologies can track the movie pirate, the video content viewed in the theater may be affected and they cannot obstruct the need of pirated movie because the watermarks in pirated moves are invisible. This paper presents a new method to defeat camcorder piracy and realize content protection in the theater using a new paradigm of information display technology, called Temporal Psychovisual Modulation (TPVM), which utilizes the differences between the human-eye perception and digital camera image-forming to stack an invisible pattern on digital screen and projector. The images formed in human vision are continuous integration of the light field, while discrete sampling is used in digital video acquisition which has "blackout" period in each sampling cycle. Based on this difference, we can decompose a movie into a set of display frames with specific patterns and broadcast them out at high speed so that the audience cannot notice any disturbance, while the video frames captured by camcorder will contain highly objectionable artifacts (i.e., the patterns). The pattern embedded in the movies can also serves as tracking information to reveal the one responsibility for the camcorder piracy.

ABSTRACT. High quality television services entail more complex systems and higher costs of prototyping, development and testing methods. The concept of Software-Defined Radio, which addresses flexibility and adaptability to new systems as they are researched, is an option to study the behavior of some technologies that will be used in future systems. This paper presents an overview of the new Advanced Television Systems Committee (ATSC) system, named ATSC 3.0. Finally, it proposes an implementation of ATSC 3.0 modulator using the GNU Radio Companion (GRC) software and validates the implementation transmitting and receiving the signal with a third party hardware and software.

ABSTRACT. With the growing popularity of a variety of devices, such as smart phones and tablets, users now have more options to watch video content, and the same content is available on different media using either broadcast or broadband. To simplify video watching for users, we have proposed Media-Unifying platform that automatically selects the appropriate delivery media according to the device function and the user's situation. This paper describes a field experiment with measuring tools that record the reception quality of both broadcast and broadband. The experiment was conducted in an environment that simulates real-life situations, such as walking or riding on a train, to verify the validity of the proposed platform. Measured data showed that the platform can increase total viewable time and reduce the amount of broadband data reception.

ABSTRACT. The provision of quality to the end users is the prime objective of the key players involved in multimedia service delivery over the Internet. Since the Internet traffic in the past decade has emerged to be more than 75\% of multimedia traffic, the issue of delivered quality to the end users is a prominent concern for both service and network providers. With the rising popularity of Over-The-Top (OTTs) services such as YouTube, Netflix, and Skype, the Internet Service Providers (ISPs) are encountering the main issue of delivering quality to the end user. Indeed, such a drastic increase of multimedia traffic requires ISPs to upgrade their networks because most of the time the quality delivered is degraded due to bottlenecks in the ISPs' networks. Moreover, within the multimedia service delivery chain, ISPs are the most affected entities by the user churn because most of the users who perceive inadequate quality switch their Internet connection to another ISP. Furthermore, the end-to-end encryption adopted by OTTs does not allow the ISPs to monitor the quality of the OTT multimedia services to apply the basic principles of network traffic management. Even if Deep Packet Inspection (DPI) may be applicable for some multimedia services which are not encrypted, user-related factors such as application, human and context would still need to be monitored somehow with additional tools for providing better and accurate information about the overall quality perceived by the users, i.e., the Quality of Experience (QoE). Hence, the ISPs must incorporate new monitoring solutions which are able to measure network parameters as well as to inspect application and context QoE influence factors. These would allow for obtaining measures closer to the QoE. In this paper, we consider the opportunity for the ISPs to incorporate passive QoE monitoring tools in the form of cloud services and user terminal probes, with the objective to provide a QoE-aware service delivery of OTT applications. The contribution of this paper is multifold. Firstly, we investigate the pros and cons of QoE monitoring at the user terminal by highlighting the roles of the different entities involved in multimedia service delivery, i.e., users, OTT and ISP, as well as by comparing this approach to other QoE monitoring approaches found in the literature. Secondly, we conducted a survey asking people specific questions regarding the monitoring at their terminals to understand privacy issues and user's preferences for the betterment of delivered quality. The analysis of the conducted survey highlights the applicability of the approach in terms of users' willingness to have probes at their terminals for QoE-aware service delivery. Thirdly, we propose a passive monitoring approach at the user terminal which considers the activation of the probes only in case quality degradation is revealed. This approach is opposite of conventional passive monitoring solutions, whose probes use to send monitored data to the data collection centers on the regular intervals (constant frequency) basis. However, this could lead to extensive utilization of computational and power resources of the user terminal. Instead, with the proposed monitoring approach, the QoE is predicted at the user terminal as a function of monitored network, application and context parameters. Depending on the service considered, specific QoE models proposed in the literature may be used for QoE prediction. Hence, the probe at the user terminal is activated only if the degradation in the quality is found beyond a certain threshold, i.e., in an event-based fashion. Thereafter, the monitored data is sent to the cloud database of the ISP, where a management system will take the optimal decisions for resolving the issue related to quality degradation over the ISP's network. Finally, we provide a comparison of the proposed approach with state of the art monitoring approaches, such as constant frequency-based monitoring approaches, on the basis of utilization of device resources such as battery life-time, RAM, and CPU as well as the amount of data generated by the monitoring probe.

ABSTRACT. The pace of technology adoption in the broadcast industry is moving forward slower than in broadband media services because of different aspects. While the broadband penetration rate is growing sharply, the required investment to embrace the broadband content catalogue into the actual broadcast solutions is a major challenge. Nowadays, broadband media services offers more content by means of Internet as the main distribution system for media exchange. Due to the success of Over-the-top (OTT) services, there is no doubt that a global transition is about to come. Nonetheless, the broadcast transition period and required investments are considerably higher than in the broadband market. The principal aim of this paper is to assess the key points to take into account to assure the compatibility of OTT content in a broadcast environment. This paper dives into the implementation considerations to make broadband purpose video processing frameworks ready for broadcast pipelines. Market solutions usually perform transcoding for legacy compatibility needing a big processing capacity while losing fidelity during recompression. This approach will generate on the fly live content usable in broadcast contexts and technical environments while saving storage, maintaining the original encoded signal when possible. The approach is a reliable and cost-effective media delivery method optimized for live HTTP-based Adaptive Streaming media and real time broadcast media delivery with muxing correction. In order to show completeness and validate the presented aspects this paper describes the performed implementation. For this purpose, professional tools of broadcast validation and reference broadband sequences have been used.