Showing papers in "IEEE Transactions on Broadcasting in 2009"

Linear Companding Transform for the Reduction of Peak-to-Average Power Ratio of OFDM Signals

Abstract: A major drawback of orthogonal frequency-division multiplexing (OFDM) signals is their high peak-to-average power ratio (PAPR), which causes serious degradation in performance when a nonlinear power amplifier (PA) is used. Companding transform (CT) is a well-known method to reduce PAPR without restrictions on system parameters such as number of subcarriers, frame format and constellation type. Recently, a linear nonsymmetrical companding transform (LNST) that has better performance than logarithmic-based transforms such as mu-law companding was proposed. In this paper, a new linear companding transform (LCT) with more design flexibility than LNST is proposed. Computer simulations show that the proposed transform has a better PAPR reduction and bit error rate (BER) performance than LNST with better power spectral density (PSD).

An Overview of IPTV Standards Development

Abstract: In the last few years, IPTV has emerged as one of the major distribution and access techniques for broadband multimedia services. It is one of the primary growth areas for the telecommunications industry. However, existing IPTV systems are generally based on proprietary implementations that do not provide interoperability. Recently, many international standard bodies have published, or are developing, a series of IPTV related standards.. This paper is an overview of the most significant recent and upcoming IPTV standards.

Subjective Assessment of Region of Interest-Aware Adaptive Multimedia Streaming Quality

Abstract: Adaptive multimedia streaming relies on adjusting the video content's bit-rate meet network conditions in the quest to reduce packet loss and resulting video quality degradations. Current multimedia adaptation schemes uniformly adjust the compression over the entire image area. However, research has shown that user attention is focused mostly on certain image areas, denoted areas of maximum user interest (AMUI), and their interest decreases with the increase in distance to the AMUI. The region of interest-based adaptive multimedia streaming scheme (ROIAS) is introduced to perform bit-rate adaptation to network conditions by adjusting video quality relative to the AMUI location. This paper also extends ROIAS to support multiple areas of maximum user interest within the same video frame. This paper presents the performance analysis of ROIAS in terms of the impact on user perceived video quality measured using subjective video quality assessment techniques based on human subjects. The tests use a wide range of video clips, which differ in terms of spatial and temporal complexity and region of interest location and variation. A comparative evaluation of both subjective and objective video quality test results is performed and demonstrate the benefit of using ROIAS for adaptive video quality delivery.

Selective Disparity Estimation and Variable Size Motion Estimation Based on Motion Homogeneity for Multi-View Coding

Abstract: Multi-view video coding (MVC) is an ongoing standard in which variable size disparity estimation (DE) and motion estimation (ME) are both employed to select the best coding mode for each macroblock (MB). This technique achieves the highest possible coding efficiency, but it results in extremely large encoding time which obstructs it from practical use. In this paper, a fast DE and ME algorithm based on motion homogeneity is proposed to reduce MVC computational complexity. The basic idea of the method is to utilize the spatial property of motion field in prediction where DE and variable size ME are needed, and only in these regions DE and variable size ME are enabled. The motion field is generated by the corresponding motion vectors (MVs) in spatial window. Simulation results show that the proposed algorithm can save 63% average computational complexity, with negligible loss of coding efficiency.

Analysis and Improvements of Zapping Times in IPTV Systems

Abstract: IPTV is becoming a serious alternative to traditional broadcast schemes, such as cable, satellite or terrestrial, to deliver TV to the home. However, the expectation of the user is that IPTV provides the same or even better quality of experience (QoE). An important part of QoE for linear TV services is the channel change time (CCT) also known as zapping time. Due to its relevance many ideas and concepts have been published which describe how to optimize or speed up the CCT. First, to be able to judge the challenges and benefits of the available concepts, a common understanding of the contributors to the CCT is necessary. This article provides an overview of all elements in the IPTV end-to-end chain influencing the CCT from the head end down to the set-top box (STB) and finally to the display. Second, various techniques for reducing the CCT are presented and discussed.

Performance-Aware Replication of Distributed Pre-Recorded IPTV Content

Abstract: Video recording in IPTV systems is a promising service that provides time-shifted services in relation to storing TV content closer to user devices such as set-top boxes. Existing approaches do not support collaboration between nodes which have correlated contents, a fact that can affect the performance of the overall system. To make this service more interactive and proactive, this paper presents the architecture using the smart personal information network (Smart PIN) as a novel performance-based content sharing network for IPTV content which uses a user-centric utility-based multimedia data replication scheme (MDRS). This allows the exchange of data based on both network performance and user interest in exchanged multimedia content in order to achieve efficient content sharing. The proposed solution is evaluated through extensive simulations and results show much improved behavior in comparison with two other existing general purpose data replication schemes.

Implications of Smoothing on Statistical Multiplexing of H.264/AVC and SVC Video Streams

Abstract: While the hierarchical B frames based scalable video coding (SVC) extension of the H.264/AVC standard achieves significantly improved compression over the initial H.264/AVC codec, the SVC video traffic is significantly more variable than H.264/AVC traffic. The higher traffic variability of the SVC encoder can lead to smaller numbers of streams supported with bufferless statistical multiplexing than with the H.264/AVC encoder (and even less streams than with the MPEG-4 Part 2 encoder) for prescribed link capacities and loss constraints. In this paper we examine the implications of video traffic smoothing on the numbers of statistically multiplexed H.264 SVC, H.264/AVC, and MPEG-4 Part 2 streams, the bandwidth requirements for streaming, and the introduced delay. We identify the levels of smoothing that ensure that more H.264 SVC streams than H.264/AVC streams can be supported. For a basic low-complexity smoothing technique that is readily applicable to both live and prerecorded streams, we identify the levels of smoothing that give (bufferless) statistical multiplexing performance close to an optimal off-line smoothing technique. We thus characterize the trade-offs between increased smoothing delay and increased statistical multiplexing performance for both H.264/AVC, which employs classical B frames, and H.264 SVC, which employs hierarchical B frames. We similarly identify the buffer sizes for the buffered multiplexing of unsmoothed H.264 SVC, H.264/AVC, and MPEG-4 Part 2 streams that give close to optimal performance.

RLS-Based Joint Estimation and Tracking of Channel Response, Sampling, and Carrier Frequency Offsets for OFDM

Abstract: This paper proposes a pilot-aided joint channel estimation and synchronization scheme for burst-mode orthogonal frequency division multiplexing (OFDM) systems. Based on the received signal samples containing pilot tones in the frequency domain, a cost function that includes the carrier frequency offset (CFO), sampling clock frequency offset (SFO) and channel impulse response (CIR) coefficients is formulated and used to develop an accompanying recursive least-squares (RLS) estimation and tracking algorithm. By estimating the channel CIR coefficients instead of the channel frequency response in the frequency domain, the proposed scheme eliminates the need for an IFFT block while reducing the number of parameters to be estimated, leading to lower complexity without sacrificing performance and convergence speed. Furthermore, a simple maximum-likelihood (ML) scheme based on the use of two long training symbols (in the preamble) is developed for the coarse estimation of the initial CFO and SFO to suppress dominant ICI effects introduced by CFO and SFO and to enhance the performance and convergence of the fine RLS estimation and tracking. Simulation results demonstrate that, over large ranges of CFO and SFO values, the proposed pilot-aided joint channel estimation and synchronization scheme provides a receiver performance that is remarkably close to the ideal case of perfect channel estimation and synchronization in both AWGN and Rayleigh multipath fading channels.

Is FM a RF-Based Positioning Solution in a Metropolitan-Scale Environment? A Probabilistic Approach With Radio Measurements Analysis

Abstract: Recent mobile devices have already contained a low-cost FM receiving function due to the continuing improvements in the device manufacturing. This paper shows that positioning based on FM signal is an alternative radio option while meeting the FCC requirement. We present a probabilistic location system using a wider-covered and longer-lived FM infrastructure. The performance is evaluated in two different metropolitan-scale environments including National Taiwan University (NTU) and Wen-Shan rural area. Both results show that the FM based location system not only satisfies the FCC requirement but also provides a comparable or even better performance to GSM based solution. Moreover, we completely analyze the realistic radio measurements of FM and GSM from four perspectives including temporal variation, spatial separation, measurement correlation and spectrum allocation. Most FM measurements are observed to provide a lower temporal variation but a weaker spatial separation than GSM. Fortunately, we discover that the lack of spatial separation can be compensated by adding additional sensed channels. This property is useful especially for a rural area where the available GSM base stations are limited and distant. Furthermore, we point out that the spatial separation of GSM signal decreases when the signal level is weaker than -90 dBm. At such a condition, FM reports a better accuracy than GSM even with the fewer channels.

Performance of Caching Algorithms for IPTV On-Demand Services

Abstract: Due to its native return channel and its ability to easily address each user individually an IPTV system is very well suited to offer on-demand services. Those services are becoming more popular as there is an undeniable trend that users want to watch the offered content when and where it suits them best. Because multicast can no longer be relied upon for such services, as was the case when offering linear-programming TV, this trend risks to increase the traffic unwieldy over some parts of the IPTV network unless caches are deployed in strategic places within it. Since caches are limited in size and the popularity of on-demand content is volatile (i.e., changing over time), it is not straightforward to decide which objects to cache at which moment in time. This paper introduces and studies a caching algorithm that tracks the popularity of objects to make intelligent caching decisions. We will show that when its parameters are set equal or close to their optimal values this algorithm outperforms traditional algorithms as LRU (least-recently used) and LFU (least-frequently used). After a generic study of the algorithm fed by a user demand model that takes the volatility of the objects into account we will discuss two particular cases of an on-demand service, video-on-demand and catch-up TV, for each of which we give guidelines on how to dimension their associated caches.

Scalable Video Coding for IPTV Services

Abstract: Scenarios for the use of the recently approved scalable video coding (SVC) extension of H.264/MPEG4-AVC in IPTV services are presented. For that, a brief technical overview of SVC when deployed in IPTV services is provided. The coding efficiency of the various scalability types of SVC is demonstrated followed by an analysis of the complexity of the various SVC tools. Based on this technical characterization, it is described how the different SVC features such as efficient methods for graceful degradation, bit rate adaptation, and format adaptation, can be mapped to application requirements of IPTV services. It is discussed how such mappings can lead to improved content portability, management and distribution as well as an improved management of access network throughput resulting in better quality of service and experience for the users of IPTV services.

Application Layer FEC for Mobile TV Delivery in IP Datacast Over DVB-H Systems

Abstract: In this paper we investigate the potential gain that can be obtained in DVB-H using application layer forward error correction (AL-FEC) to perform a multi-burst protection of the transmission for improving the reception of streaming services for mobile terminals. Compared to the conventional approach with link layer multi protocol encapsulation FEC (MPE-FEC), this technique allows to increase the robustness of the DVB-H transmission not only as a function of the capacity devoted for error repair (FEC overhead), but also as a function of the number of bursts jointly encoded. The main drawback of this approach is an increase of the network latency, that can be translated into a larger service access time, and, in the case of mobile TV, a larger zapping time between channels, which is currently seen as a crucial parameter for DVB-H usability. In this paper the performance of the proposed approach is evaluated using field measurements. We evaluate the gain compared to MPE-FEC in terms of reduced IP packet error rate of the streaming service as a function of the FEC overhead and the latency introduced. Moreover, simulations have been performed to quantify feasible link margin gains. We also discuss techniques to reduce and hide the zapping time from the perception of the users.

Channel Estimation Based on a Time-Domain Threshold for OFDM Systems

Abstract: Channel estimation for OFDM systems is usually carried out in frequency domain by the least-squares (LS) method using known pilot symbols. The LS estimator has a merit of low complexity but may suffer from noise because it does not consider any noise effect in obtaining its solution. To enhance the noise immunity of the LS estimator, we consider the estimation noise in time domain named discrete Fourier transform (DFT)-based channel estimation. Residual noise existing at the estimated channel coefficients in time domain could be reduced by reasonable selection of a threshold value. To achieve this, we propose a channel-estimation method based on a time-domain threshold which is a standard deviation of noise obtained by wavelet decomposition. Computer simulation shows that the estimation performance of the proposed method approaches to that of the known-channel case in terms of bit-error rates after the Viterbi decoder in overall SNRs.

Preserving Video Quality in IPTV Networks

Abstract: Growing numbers of service providers are implementing video services over IP networks, and discovering the unique challenges of providing a high quality-of-experience (QoE) to subscribers. Delivering consistent QoE in packet-switched networks can be a complex proposition due to the high sensitivity of video traffic to packet loss, as well as to delay and jitter. Preserving video quality in IP Television (IPTV) networks that mostly rely on copper access lines poses an even greater challenge. Service providers need intelligent mechanisms in core and distribution networks to prevent congestion that deteriorates video quality, as well as intelligence between the aggregation networks and IP set-top box (STB) to repair packet losses, speed up channel change times and monitor the quality of the subscriber experience. This paper discusses a series of core and aggregation-layer approaches, collectively referred to as Visual Quality of Experience (VQE) to address these issues. The VQE approach encompasses on-path video connection admission control (CAC) employing Resource ReSerVation Protocol (RSVP) in the core and aggregation layers, and a real-time signaling mechanism operating between the provider aggregation edge and the IP STBs to address packet losses, long channel change times and quality monitoring.

Adaptive Hybrid Transmission Mechanism for On-Demand Mobile IPTV Over WiMAX

Abstract: In this paper, we propose an Adaptive Hybrid Transmission (AHT) scheme for on-demand mobile IPTV service over broadband wireless access network (i.e. mobile WiMAX, 802.16e). Proposed algorithm utilizes hybrid mechanism which combines multi-channel multicasting and unicast scheme to enhance not only service blocking probability but also reduce overall bandwidth consumption of the wireless system which has very limited resources compared to wired networks. An adaptive resource allocation algorithm is also proposed, and is shown to achieve minimum blocking probability. In order to evaluate the performance, we compare proposed algorithm against traditional unicast and multicast schemes.

Low Complexity Iterative ICI Cancellation and Equalization for OFDM Systems Over Doubly Selective Channels

Abstract: In this paper, a low complexity iterative intercarrier interference (ICI) cancellation and equalization technique is proposed for use in OFDM systems over doubly selective channels. In the iterative parallel interference cancellation/minimum mean square error (PIC/MMSE) detector has a high complexity and a restriction on the structure which can not remove the ICI in the initial stage. Therefore, an error propagation occurs due to the ICI regenerated by the incorrect output of soft-input soft-output (SISO) decoder. In order to reduce the error propagation, an MMSE detector based on the successive interference cancellation (SIC) is used in the initial stage. The low complexity MMSE detector is also derived to minimize the error propagation by quantifying the decision error before SISO decoding. In the first iteration, simulation results show that the proposed scheme outperforms the conventional PIC/MMSE scheme by about 3 dB at bit error rate (BER)=1 times 10-3 while maintaining the equivalent computational complexity. In the subsequent iteration, it is possible to cancel the ICI out in the received signals by the aid of soft log-likelihood ratio (LLR) fed from the SISO decoder. Converting the LLR to the decision error probability, the error covariance matrix is obtained more accurately. As a result, the error propagation can be effectively reduced by dealing with only the dominant components, when considering decision errors. Finally, simulation results show that the proposed scheme outperforms the conventional PIC/MMSE scheme.

A Novel Multi-Points Square Mapping Combined With PTS to Reduce PAPR of OFDM Signals Without Side Information

Abstract: In this paper, we first propose a novel multi-points square mapping (MSM) scheme Then, we describe in detail how to combine the proposed MSM scheme with conventional partial transmit sequence (C-PTS) scheme, named as M-PTS, to reduce the PAPR of OFDM signals Compared with C-PTS, the proposed M-PTS scheme needs not to submit side information while keeping almost the same performance of PAPR reduction as the C-PTS scheme Extensive simulations are conducted to validate analytical results, showing that the proposed M-PTS scheme has better bandwidth efficiency and bit error ratio (BER) performance compared with the C-PTS scheme

SASHA—A Quality-Oriented Handover Algorithm for Multimedia Content Delivery to Mobile Users

Abstract: The convergence of the existing network access technologies to a common IP-based architecture and the increase in popularity of accessing video content over the Internet makes IPTV a promising solution for media and entertainment industries. Additionally, video content delivery to the increasingly popular mobile devices over heterogeneous wireless networks makes IPTV even more appealing. However the distribution of multimedia content over heterogeneous wireless networks to mobile devices involves significant technical challenges related to mobility management and quality of service provisioning. The existing solutions do not consider quality of service as a decision making parameter for mobility management in general and handover management in particular. This paper proposes the smooth adaptive soft handover algorithm (SASHA), a novel quality-aware approach to handover based on load balancing among different networks using a comprehensive, quality of multimedia streaming (QMS), function for decision making. SASHA represents the handover management solution at the core of the more comprehensive multimedia mobility management system (M3S), a quality oriented mobility management framework for multimedia applications which maximizes user perceived quality by efficiently exploiting all available communication resources. Simulation-based testing results are presented, outlining the performance of SASHA in different mobility scenarios. The evaluation is performed for different number of nodes performing handover simultaneous and for various situations in terms of networks' overlapping area. The results shown indicate how SASHA outperforms other three mobility management solutions in terms of quality, scalability and resilience to the dynamics of the networks' overlapping area.

IPTV Multicast Over Wireless LAN Using Merged Hybrid ARQ With Staggered Adaptive FEC

Abstract: Internet Protocol TV (IPTV), an emerging Internet application, provides more flexibility and interactivity for users because of its embedded return channel. Wireless 802.11 networks, used as the last hop of IPTV networks, would add another dimension of mobility, which is of great value for hot spot deployment and is increasingly required for in-home content distribution. However, a wireless channel may suffer from multi-path fading and interference, which may cause random and burst packet loss and impact users' quality of experience of an IPTV program. We report the design, implementation and evaluation of merged hybrid adaptive FEC and ARQ (MHARQ) system for IPTV multicast over wireless LANs. In MHARQ, an IPTV cache server is introduced to improve the reliability for the last hop wireless distribution. MHARQ combines the advantages of receiver-driven staggered FEC and hybrid ARQ schemes to compensate the large dynamic range of WLAN channels and to achieve high reliability, scalability and wireless bandwidth efficiency for IPTV multicast. We also address backward compatible FEC encoding and decoding, audio and video synchronization issues in practical system implementation. Using the ORBIT radio grid test bed, we have investigated the performance of the proposed MHARQ system with various numbers of users per AP and different numbers of APs per IPTV cache server. It is demonstrated via real system implementation on ORBIT that MHARQ improves wireless bandwidth efficiency and scalability for reliable IPTV multicast, compared with existing reliable multicast schemes.

Power Alignment of Digital Predistorters for Power Amplifiers Linearity Optimization

Abstract: In this paper, a study of the power alignment issue in digital predistorters is presented. The proper alignment is achieved by adjusting the normalization gain used to synthesize the predistortion function. The dependencies of the linearity and power efficiency of the linearized amplifier upon the gain normalization factor are investigated, and it is shown that the efficiency of the linearized amplifier is almost unaffected by variation of the normalization gain. Conversely, the linearity performance of the linearized power amplifier is found to be dependent on the gain normalization factor, as a consequence of the average power variation through the predistorter. Indeed, the proper power alignment of the predistorter following an adequate choice of the normalization gain shows a significant improvement in the measured adjacent channel power ratio at the linearized amplifier output.

MAC Layer Multicast Error Control for IPTV in Wireless LANs

Abstract: Internet Protocol Television (IPTV) over IEEE 802.11 Wireless LANs, which are considered today as the de-facto wireless access network for local distribution (home networks, hotspots), brings forth a big challenge to guarantee high quality multicast/broadcast delivery over unreliable and time-variant wireless channels. However, in IEEE 802.11 Wireless LANs, current standard MAC layer protocols do not provide any error correction scheme for broadcast/multicast. In our previous work, we enhanced a Leader Based Protocol (LBP) and proposed a Beacon-driven Leader Based Protocol (BLBP) for the MAC layer multicast error control. However, as pure Automatic Repeat reQuest (ARQ) schemes, both LBP and BLBP are not efficient for large multicast groups. In this paper, we combine BLBP and packet level Forward Error Correction (FEC) and propose a Hybrid Leader Based Protocol (HLBP) for the MAC layer multicast error control. HLBP transmits the original data packets using raw broadcast and retransmits parity packets using an improved BLBP which is based on block feedback. To guarantee the required Packet Loss Ratio (PLR) under strict delay constraints for multicast delivery over a Gilbert-Elliott (GE) channel, we analyze LBP, BLBP and HLBP, develop analytic performance models that allow optimizing the configurations. The performance models are verified via simulation experiments. Both the theoretical analysis and simulation results show that HLBP is much more efficient than LBP and BLBP especially for large multicast groups and is also more efficient than the best application layer multicast error correction scheme. BLBP needs the minimum number of redundancy transmissions among all pure ARQ based schemes while HLBP needs the near-minimum number of redundancy transmissions among all schemes. Due to the simplicity and efficiency, BLBP is a good choice for IPTV multicast delivery for small groups in Wireless LANs while HLBP is a better choice for large multicast groups.

System Scheduling for Multi-Description Video Streaming Over Wireless Multi-Hop Networks

Abstract: Providing real-time multimedia applications over wireless multi-hop networks is a challenging problem because the wireless channels are highly sensitive to delay, interference and topology changes. Multiple description coding (MDC), as a new emerging error-resilient technique, has been widely used recently in wireless video transmission. Its fundamental principle is to generate multiple correlated descriptions such that each description approximates the source information with a certain level of fidelity. Inevitably, MDC introduces many description streams which may influence each other and thus, reasonable system scheduling is needed to provide a satisfied video quality. The novelty of this work is to investigate the optimal distributed scheduling for multiple competing MDC streams in a resource-limited wireless multi-hop network. This is achieved by joint optimization of MDC, rate control and multipath routing. Two joint optimal algorithms, namely a distributed rate control and routing (DRCR) and a simplified DRCR algorithm, are proposed to solve this problem with constraints that arise from the multiple description streams among multiple users via multiple paths. Both algorithms are designed in a distributed manner that is amenable to on-line implementation for wireless networks. Theoretical analysis and simulation results are provided which demonstrate the effectiveness of our proposed joint schemes.

Distributed Storage-Assisted Data-Driven Overlay Network for P2P VoD Services

Abstract: Providing VCR-like operations in peer-to-peer (P2P) environments is a significant challenge. This paper proposes a distributed storage-assisted data-driven overlay network (SDNet) to support P2P video-on-demand (VoD) services. It integrates two networks: a data-driven overlay network (DONet) and a multi-way tree. DONet is enhanced and used for the routine video distribution based on the buffer overlapping mechanism and gossip protocol. A novel algorithm which uses a multi-way tree structure and extra pre-fetching buffers at the nodes is proposed to support efficient VoD operations. Videos are divided into uniform segments, pre-fetched and stored in a distributed manner along the tree topology. The cooperation between DONet-based video delivery and the tree-located multimedia components enable multimedia streaming interactive commands to be performed efficiently. This paper presents and discusses the structure of SDNet and the distributed storage scheme and details the cooperation procedure. Simulation-based testing results show how the proposed SDNet is an efficient interactive streaming solution in a P2P environment.

Efficient Non-Pilot-Aided Channel Length Estimation for Digital Broadcasting Receivers

Abstract: Channel estimation and equalization techniques are crucial for the ubiquitous broadcasting systems. Conventional receivers for most broadcasting or wireless standards preset the channel length to the maximal expected duration of the channel impulse response for the adopted channel estimation and equalization algorithms. The excessive channel length often significantly increases the implementational complexity of the wireless receivers and leads to the redundant information which would induce the additional estimation errors. Moreover, such a scheme does not allow the dynamic memory allocation for variable channel lengths. This could further increase the power consumption and reduce the battery life of a mobile device. The knowledge of the actual channel length would, in principle, help the system designers decrease the complexity of the channel estimators using maximum likelihood (ML) and minimum-mean-square-error (MMSE) algorithms. In this paper, we address this important channel length estimation problem and propose a novel autocorrelation-based algorithm to estimate the channel length without the need of pilots or training sequence. The associated threshold for the channel length estimation depends on the sample size, the signal-to-noise ratio and the leading/last channel coefficients. In addition, we provide the mean-square analysis on the effectiveness of the proposed non-pilot-aided channel length estimator through Monte Carlo simulations.

A Broadcasting Scheme for Selective Contents Considering Available Bandwidth

Abstract: 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.

Robust and Accurate Frequency and Timing Synchronization Using Chirp Signals

Abstract: We propose a new robust and accurate synchronization procedure using a training sequence composed of chirp signals. We use a new integer frequency estimation algorithm and propose a new combination of a known fractional frequency offset estimation algorithm and timing synchronization algorithm. The training sequence is composed of one up and two down chirp symbols, also known as Newman phases. The integer frequency offset estimation algorithm uses the effect of timing and frequency offsets on the matched filter outputs of the chirp signals. Autocorrelation and reversed autocorrelation are used to acquire the timing instant and the fractional frequency offset. We present the complete timing and frequency synchronization procedure and study the output signals of the autocorrelation and reversed autocorrelation algorithms. Finally, we check the performance of the synchronization procedure via Monte Carlo simulation in several multipath channels. Our algorithms are accurate and more robust compared to previously published state-of-the art algorithms.

Extensive Penetration Loss Measurements and Models for Different Building Types for DVB-H in the UHF Band

Abstract: DVB-H networks allow high data rate broadcast access for hand-held terminals. Building penetration loss measurements of a DVB-H signal at 602 MHz are performed in 100 buildings in Belgium. Buildings are categorized in different types (office buildings, apartments, villas, mansions, row houses, stations) and the rooms are classified according to the number of outside radiated walls. The cumulative distribution function of the penetration loss is determined and lognormality is investigated. Models are developed to calculate the penetration loss as a function of the number of radiated walls. Also models for apartments as a function of the floor level are developed. This research enables the calculation of indoor coverage probability for wireless networks.

Internet Multicasting of IPTV With Essentially-Zero Delay Jitter

Abstract: A technology for multicasting packetized multimedia streams such as IPTV over the Internet backbone is proposed and explored through extensive simulations. An RSVP or DiffServ algorithm is used to reserve resources (i.e., bandwidth and buffer space) in each packet-switched IP router in an IP multicast tree. Each IP router uses an Input-Queued (IQ) switch architecture with unity speedup. A recently proposed low-jitter scheduling algorithm is used to pre-compute a deterministic transmission schedule for each IP router. The IPTV traffic will be delivered through the multicast tree in a deterministic manner, with bounds on the maximum delay and jitter of each packet (or cell). A playback buffer is used at each destination to filter out residual network jitter and deliver a very low-jitter video stream to each end-user. Detailed simulations of an IPTV distribution network, multicasting 75 high-definition video streams over a fully-saturated IP backbone are presented. The simulations represent the transmission of 129 billion cells of real video data and where performed on a 160-node cluster computing system. In the steady-state, each IP router buffers approx. 2 cells (128 bytes) of video data per multicast output-port. The observed delay jitter is zero when a playback buffer of 15 milliseconds is used. All simulation parameters are presented.

DVB-T/H and T-DMB: Physical Layer Performance Comparison in Fast Mobile Channels

Abstract: 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 antennas. 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.

Content and Service Protection for IPTV

Abstract: 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.