Immersive communications is a key use case for 6G where applications require reliable latency-bound media traffic at a certain data rate to deliver an acceptable User Experience (UX) or Quality-of-Experience (QoE). The Quality-of-Service (QoS) framework of current cellular systems (4G and 5G) and prevalent network congestion control algorithms for latency-bound traffic like L4S typically target network-related Key Performance Indicators (KPIs) such as data rates and latencies. Network capacity is based on the number of users that attain these KPIs. However, the UX of an immersive application for a given data rate and latency is not the same across users, since it depends on other factors such as the complexity of the media being transmitted and the encoder format. This implies that guarantees on network KPIs do not necessarily translate to guarantees on the UX. In this paper, we propose a framework in which the communication network can provide guarantees on the UX. The framework requires application servers to share real-time information on UX dependency on data rate to the network, which in turn, uses this information to maximize a UX-based network utility function. Our framework is motivated by the recent industry trends of increasing application awareness at the network, and pushing application servers towards the edge, allowing for tighter coordination between the servers and the 6G system. Our simulation results show that the proposed framework substantially improves the UX capacity of the network, which is the number of users above a certain UX threshold, compared to conventional rate control algorithms.

DASH has become one of the widely adopted and contributed streaming formats for internet streaming services. DASH Industry Forum has been the core catalyzer in this with interoperability discussions supported by publicly available Interoperability Guidelines and testing tools. This paper introduces DASH-IF activities and its tools and proposes a demonstration based on the recent study of low latency DASH service support.

DASH content delivery via ROUTE is being standardized within ATSC and is supported by the interoperability efforts within DASH-IF. The goals achieved by this work include effectively demonstrating the key technological features of the nascent standards on a variety of platforms. These features include demonstration of hybrid services delivery and customized advertisement insertion, etc. At the same time, this demonstration is accomplished using open-source tools, content, and components to help foster harnessing and utilizing the full potential of these technologies in the industry. All of this is supplemented by near-product grade stable and optimized performance demonstration.

This special issue is concerned with the latest developments in state-of-the-art adaptive media streaming technologies and applications.

This paper provides an overview on the challenges of mobile video streaming focusing on overload situations, variable bandwidth and power consumption issues. Dynamic Adaptive Streaming over HTTP (DASH) is introduced as an enabler to address a significant portion of these challenges. The applicability of the formats to 3G and 4G unicast distribution is provided. Also the applicability of DASH formats for multicast distribution in MBMS is shown. Finally, the usage of DASH in hybrid unicast/multicast is introduced to provide a flexible extension to support different business and delivery models. An outlook to ongoing and future work is provided.

This paper proposes an adaptive HTTP-based video streaming framework for mobile networks using the Scalable Video Coding (SVC) extension to the H.264 standard. We present a method to statistically estimate the channel at the mobile client and use it in our work to adapt the bit rate of the video. The adaptation takes into account both the quality contributions and the probability of successful timely decoding for different video segments. The simulation results show significant improvements in terms of a reduction of playback interruptions and improved perceived quality of service.

In this paper, we provide some insight and background into the Dynamic Adaptive Streaming over HTTP (DASH) specifications as available from 3GPP and in draft version also from MPEG. Specifically, the 3GPP version provides a normative description of a Media Presentation, the formats of a Segment, and the delivery protocol. In addition, it adds an informative description on how a DASH Client may use the provided information to establish a streaming service for the user. The solution supports different service types (e.g., On-Demand, Live, Time-Shift Viewing), different features (e.g., adaptive bitrate switching, multiple language support, ad insertion, trick modes, DRM) and different deployment options. Design principles and examples are provided.

The first two generations of TV broadcasting are almost history, at least from the standardization perspective. Analog and digital satellite distributions both have addressed mass markets. The next-generation TV systems are close to primetime deployments: HDTV, 3DTV, interactive services, hybrid services, and additional innovations will dominate the next-generation TV. With the efficiency and maturity of DVB-S2, satellite distribution is a cost-efficient and well-established broadcast technology with significant potential for extensions. In this paper, the future perspectives of digital TV and HDTV broadcasting will be first explored, considering ongoing and future standardization activities that will be carried out. Particular attention to innovative solutions based on adaptive modulation and coding, source and channel coding, and error resilience techniques for satellite TV transmission is paid. In addition to broadcast TV, also the perspectives of hybrid and IPTV will be considered in a satellite scenario with their pros and cons, trying to understand if satellite IPTV will be in competition with “conventional” broadcast satellite TV services (like it already happens in terrestrial scenarios).

This paper presents a novel adaptive video streaming framework designed for OFDM-based mobile networks. The proposed adaptation algorithm exploits the benefits of the Scalable Video Coding (SVC) extension of the H.264/AVC standard together with cross-layer information from the mobile terminals in performing adaptation that maximizes streaming qualities amongst all participating users. Simulations for adaptive video streaming along with the well-known TFRC-based and no adaptation scenarios were conducted in a realistic mobile network simulator. Our results show significant improvements in all aspects for the introduced framework over both the TFRC-based and no adaptation cases.

This paper investigates the performance of TCP-Friendly Rate Control (TFRC) to control the transmission rate of scalable video streams when used in a mobile network. The streams are encoded using the Scalable Video Coding (SVC) extension of the H.264/AVC standard. Adding or removing the layers is decided based on the TFRC during varying channel conditions of the mobile network. We conduct simulations in various realistic use cases, evaluate and compare the performance with and without TFRC-based adaptation. The results show significant improvements in terms of lower loss rate, delay, required buffer size and less playback interruption.

Dynamic interactive multimedia scenes (DIMS) is the latest 3GPP multimedia standard aiming at the development and delivery of rich media services over mobile networks, specifically for the constrained devices. DIMS is a concept that encompasses a declarative format for audio-visual scene representation, as a composition of rich media objects with specific properties and behaviour. This paper focuses on the issues related to the delivery of Interactive TV like services with DIMS in mobile broadcast environments. We have investigated the system design parameters which would ensure better QoS for the subscribers. With the simulation results, we have setup guidelines for the selection of application layer FEC parameters, video coding settings and optimum base station transmit power. These guidelines assist in improving the overall rich media experience offered to the end-users.

Error robust video communication for hand-held devices is a delicate task because of limited computational resources and hostile channel conditions in mobile environments. The loss of coded video data on the channel can result in spatio-temporal error propagation in the video. In addition, stringent end-to-end delays for conversational applications make this challenge even more difficult. In this work, we investigate several techniques which exploit feedback from the receiver to enhance the performance of conversational video in realistic mobile communication environments. Specifically, we show how a low-complexity interactive error tracking technique can be combined with multiple reference picture selection (RPS) based on the existing syntax of H.264/AVC. This technique outperforms other interactive error protection strategies by a margin of more than 2 dB for moderate channel loss rates, with minimal impact on end-to-end delays.

Recent advances in forward error correction and scalable video coding enable new approaches for robust, distributed streaming in mobile ad hoc networks (MANETs). This work presents an approach for distribution of real time video by different uncoordinated peer-to-peer relay or source nodes in an overlay network on top of a MANET. The approach proposed here allows for distributed, rate-distortion optimized transmission-rate allocation for competing scalable video streams at relay nodes in the overlay network. Furthermore the approach has the desirable feature of path/source diversity for enhancing reliability in connectivity to serving nodes. Signaling overhead within the overlay network is kept at a minimum, since optimizations are done at relay nodes and clients rather than at servers.

Robust video conversational applications for hand-held devices come with numerous challenges, e.g. real-time processing, complexity constrained devices and small end-to-end delays, etc. Transmission losses of compressed video data result in spatio-temporal error propagation in the decoded video sequence. To ensure some QoS, the video codec has to be well tuned to combat the degradation resulting from losses. Several feedback based error mitigation technique are assessed in this work. The proposed error robustness technique based on reference picture selection (RPS) and error tracking enhances the overall performance of the target system by more than 4 dB for moderate radio link control (RLC) PDU loss rates of 1.5%. This enhancement is achieved without any additional computational complexity.

The scalable video coding (SVC) standard as an extension of H.264/AVC allows efficient, standard-based temporal, spatial, and quality scalability of video bit streams. Scalability of a video bit stream allows for media bit rate as well as for device capability adaptation. Moreover, adaptation of the bit rate of a video signal is a desirable key feature, if limitation in network resources, mostly characterized by throughput variations, varying delay or transmission errors, need to be considered. Typically, in mobile networks the throughput, delay and errors of a connection (link) depend on the current reception conditions, which are largely influenced by a number of physical factors. In order to cope with the typically varying characteristics of mobile communication channels in unicast, multicast, or broadcast services, different methods for increasing robustness and achieving quality of service are desirable. We will give an overview of SVC and its relation to mobile delivery methods. Furthermore, innovative use cases are introduced which apply SVC in mobile networks.

Abstract The integration of reliable Video‐on‐Demand (VoD) broadcasting schemes in mobile datacast systems, specifically in DVB‐H, is studied and enhanced. Sophisticated VoD broadcasting schemes such as Harmonic Broadcasting (HB) allows receivers to tune into the ongoing transmission of a video stream at arbitrary time, while still being able to receive the multimedia sequence from beginning to end, after short initial playout latency. In addition, we address service enhancements by using scalable video coding (SVC) to support heterogeneous receiver capabilities and receiving conditions as well as the reception of the signal from more than on transmission site. We present and discuss options for the integration of VoD broadcasting schemes in combination with fountain codes. Optimizations in parameter selection are discussed. A realistic protocol environment is only slightly modified to support our system concept. Simulation results show the benefits of the discussed VoD scheme compared to existing approaches if integrated in DVB‐H. Copyright © 2007 John Wiley & Sons, Ltd.

Abstract — 3GPP has specified Multimedia Broadcast/Multicast Services (MBMS) addressing the increasing demands of multimedia download and streaming applications in mobile scenarios. Thereby, video coding is an essential component and H.264 is the only recommended video codec in MBMS mainly due to its high compression efficiency and easy network integration. In this paper we introduce the main system design parameters that influence the performance of MBMS video streaming over EG-PRS and UMTS using H.264 encoded streaming. Effective design methodology that includes robustness against packet losses and efficient use of the scarce radio resources is presented. Thereby, care is taken also about the limited processing power of mobile terminals, service delay constraints, and heterogeneous receiving conditions. We also investigate an advanced receiver concept, the so called permeable–layer receiver. Selected simulation results show the suitability of certain parameter settings as well as the benefits provided by the advanced receiver concept. I.

In this work we address reliablefile delivery over mobile broadcast networks, concentrating on the Raptor codes as specified for Multimedia Broadcast/Multicast Services (MBMS) within 3GPP. We start by describing Luby-Transform (LT) codes, which are the first practical fountain codes. Then, using a natural and easy to understand linear algebra notation, we describe Raptor codes as a powerful extension of LT codes. We provide some insight into the Raptor code structure and some guidelines for implementation of encoders and decoders. Finally, some selected simulations verify the good performance of file distribution with Raptor codes as specified in 3GPP. References to a complete set of simulation results are also provided. I. INTRODUCTION

Raptor codes have been recently standardized by 3GPP to be used in the application layer. These codes are very efficient and perform sufficiently close to the so-called ideal fountain code on the block-erasure channel. However, for real applications, a simple receiver enhancement called permeable layer receiver can be done, which provides huge performance gains. Standard Raptor decoding algorithms have to be revised and new efficient decoding algorithms need to be introduced. In this work we present a practical and efficient implementation of the Raptor decoder for permeable layer receiver. We show that, not only we achieve huge performance gains, but we achieve them at an affordable (low) decoding complexity. Complexity and decoding performance simulation results are presented for the download delivery and video broadcasting.

Video transmission over networks is getting increasingly important, thus upcoming network types like mobile ad-hoc networks (MANETs) can also become a suitable platform for exchanging/sharing real-time video streams. We present an approach using different sources with linear independent representations of a layered video stream for increasing the robustness in transmission. This approach is based on the scalable video coding (SVC) extensions of H.264/MPEG-4 AVC with different layers for assigning importance for transmission. Additionally a novel unequal packet loss protection (UPLP) scheme based on Raptor forward error correction codes is employed. This scheme allows for reception from different sources and benefits from it. While the reception of a single stream guarantees base quality at least, the combined reception enables play-back of video of full quality and/or lower error rates.

We propose a dynamic sharing of radio resources in a wireless multiuser system by combining a flexible video coding scheme, namely the currently developed scalable video coding (SVC) extension of H.264/AVC with appropriate radio link buffer management for multiuser streaming services. The paper introduces the components of the proposed system and the used cross-layer interface. Simulation results verify the benefits of our approach

Third Generation Partnership Project (3GPP) and Digital Video Broadcasting (DVB) have just recently specified the use Raptor codes in their mobile broadcast file delivery engine. Until today, investigations of the applicability of these codes to the mentioned systems have been carried out by assuming almost exclusively quite simple loss models such as statistically independent radio packet losses. Furthermore, the combination and tradeoffs between physical layer parameters such as transmit power or physical layer for-ward error correction code rates in system design has been completely ignored. In this work we investigate the end-to-end system performance by analyzing the trade-off between physical layer parameters and application layer code. In particular we are interested in determining the optimal system operating points for optimized broadcast file delivery. Parameters such as power allocation, mobility model, physical layer Turbo code rate, raptor code expansion ratio are investigated and a single performance criteria is defined, namely the required energy to deliver a file to at least some high percentage of users. It is shown that typically considered system configurations with high transmit power and low physical layer code rates - resulting in low radio block loss rates - are in general suboptimal for efficient delivery. A careful tradeoff the system parameters can reduce the required resources in the order of a magnitude.

We consider wireless broadcasting of multimedia content to allow asynchronous media access. Receivers subscribe at any time to the ongoing broadcast session, but are still able to display the media stream from the beginning. A fully scalable broadcasting scheme is presented where the media stream is appropriately segmented and segments are protected by fountain codes. The decoding behavior of rateless codes on wireless as well as on erasure channels is considered within the framework of information collection. Asynchronous data access and full reliability at the same time are achieved. Depending on its receiving conditions the receiver adapts its initial playout delay for high probability of successful playout. Analytical expressions for the failure probability of successful media playout are derived depending on the initial delay and the channel conditions at the receiver.

In this work we show the suitability of H. 264/AVC extended profile for wireless video streaming applications. In particular, we exploit the advanced bitstream switching capabilities of H. 264/AVC using SP/SI pictures. Optimized encoders for switching pictures are proposed. Finally, a framework for dynamic switching and frame scheduling is proposed and the performance is shown for H. 264/AVC video streaming over EGPRS.

In this paper different means for adding reliability in different protocol layers are discussed, in order to reliably broadcast multimedia data within existing wireless networks. In particular, we investigate the performance of additional forward error correction in the physical layer, the radio link control layer, and the RTP layer for multimedia broadcast and multicast services over GERAN systems. Advantages and drawbacks when applying means for reliability in different layers are shown. We introduce a simple receiver modification, referred to as permeable-layer receiver (PLR), which exploits traditionally useless information at the receiver, while the transmitter is kept unchanged. Significant performance gains are reported. Finally, the application to H.264/AVC based wireless video broadcast is discussed and the performance of different system designs for video transmission is shown.

Various advanced methods have been proposed to achieve sufficiently high throughput as well as reliability for multimedia broadcast multicast services (MBMS) within 3GPP RAN and GERAN. One of the methods which yield promising results with only marginal system modifications is based on outer Reed-Solomon (RS) coding on the radio link control (RLC) layer. For initial studies standard decoding algorithms based on erasure-only decoders have been employed. In order to further improve the performance, the system can be viewed as a serially concatenated code such that iterative decoding can be performed. Then, advanced receivers for MBMS over GERAN would incorporate soft-in soft-cut (SISO) decoders for both the inner convolutional code, as well as the outer RS code. Whereas for the convolutional code, standard SISO algorithms are applied, for the outer RS code a standard Berlekamp-Massey algorithm is modified to accept soft-input and to provide soft-output. With this modified iterative decoding strategy, simulations have shown gains of up to 14% in throughput on typical mobile radio channels.

In this work, reliable retransmission strategies, including point-to-point (p-t-p), point-to-multipoint (p-t-M) and point-to-multipoint incremental redundancy (p-t-M-IR) are studied with special focus on their applicability to multimedia broadcast and multicast services (MBMS) over GERAN. By employing a binary OR multiple access channel (BORMAC) model, analytical expressions for the system throughput, the residual RLC/MAC block loss rate, and the residual IP-packet loss rate are derived for all investigated schemes. Furthermore, a simple NACK oriented feedback policy, which allows the utilization of a resource efficient feedback channel in cellular environments is presented. It is shown that the p-t-M-IR scheme outperforms both the p-t-p and the p-t-M retransmission scheme significantly in terms of system throughput, residual loss RLC/MAC block loss rate, as well as residual IP-packet loss rate. The throughput reduction of the p-t-M-IR scheme with increasing number of mobiles in the serving area is significantly less than with other ARQ schemes.

In this paper, forward error correction (FEC) on the application layer for reliable multicast transmission over wireless is investigated. A wireless channel model is introduced and analytical performance bounds for FEC on the application layer over wireless are given. It is shown that with state-of-the-art receiver strategies the residual application layer packet loss rate exceeds the required target loss rates and that sufficient quality-of-service (QoS) cannot be achieved. A simple receiver modification, called permeable-layer receiver (PLR), is introduced, which exploits traditionally useless information at the receiver, while the transmitter is kept unchanged. The PLR is shown to outperform the traditional receiver, allowing the efficient support of many wireless receivers in a multicast environment at the same time. Simulation results with LDGM staircase codes are provided, being consistent with theoretical findings. Future work items to support and extend the conceptual findings are discussed.

In this work we investigate the performance of the H.264/AVC video coding standard for low bit rate mobile multimedia messaging services (MMS). We focus on the appropriate selection of the quantization parameter and the temporal resolution. For this purpose, a psychovisual experiment has been designed. It is revealed that only limited set of parameter values is necessary to span almost the entire range of quality levels. In general quantization parameters of 34 frame rates of 10 fps and bit rates below 64 kbit/s are sufficient to provide good quality. Sports sequences ask for slightly higher frame rate, 15 fps is sufficient. To provide sufficient quality, quantization parameters below 40 and frame rates below 5 fps should definitely not be used.

In this work we compare nonscalable video coding with data partitioning using H.264/AVC under similar application and channel constraints for conversational applications over mobile channels. For both systems optimized rate allocation and network feedback has been applied. From the experimental results it is observed that based on the average PSNR the nonscalable system outperforms the data partitioning system. However, with the data partitioning system the percentage of entirely lost frames can be lowered, and the probability of poor quality decoded video can be reduced.

In this work, we investigate the integration of H.264/AVC based video into the multimedia broadcast/multicast service (MBMS). MBMS allows simultaneous distribution of live video with reasonable quality to several or even a large number of users within a serving area. We discuss several system design options and propose a simple, but nevertheless efficient cross-layer design concept, which helps to determine an optimal set of both application and transmission parameters for maximizing the quality at the user terminals for different link conditions. In order to validate our concept, we present detailed performance results using realistic test conditions provided by 3GPP.

We consider streaming of video sequences over both constant and variable bit-rate (VBR) channels. Our goal is to enable decoding of each video unit before exceeding its displaying deadline and, hence, to guarantee successful sequence presentation even if the media rate does not match the channel rate. In this work, we show that the separation between a delay jitter buffer and a decoder buffer is in general suboptimal for VBR video transmitted over VBR channels. We specify the minimum initial delay and the minimum required buffer for a given video stream and a deterministic VBR channel. In addition, we provide some probabilistic statements in case that we observe a random behavior of the channel bit rate. A specific example tailored to wireless video streaming is discussed in greater detail and bounds are derived which allow guaranteeing a certain quality-of-service even for random VBR channels in a wireless environment. Simulation results validate the findings.

A general joint source-channel rate allocation scheme for a dependent video coding environment is presented. It is assumed that the source coders rely on motion-compensated prediction, and thus error propagation will contribute significantly to the degradation of the reconstructed video. The rate allocation methodology is based upon generation of operational distortion-rate characteristics. In order to reduce computational complexity, these surfaces and the channel code performance are modeled. An analytic method for computing optimal rate allocation across frames in a video sequence is then introduced. Comparisons are made between equal error protection and unequal error protection over video frames. Results are shown for H.263 compressed video used in conjunction with rate-compatible punctured convolutional codes.

Standard compliant enhancements of the emerging H.264/AVC coding algorithm for transmission over lossy IP-based networks are described with the error resilience features being presented in greater detail. Standard compliant encoder and decoder enhancements as well as the exploitation of transport layer mechanisms to improve the quality in packet lossy environments are presented. Different schemes are compared and appropriate experimental results based on common test conditions are discussed. Finally, the selection of suitable features is discussed.

In this work we compare standard-compliant H.264 video transmission with an efficient fine granular scalable coding method based on H. 264, the progressive texture video coding (PTVC) scheme in a wireless conversational application environment. Therefore, we operate on a simple but meaningful wireless channel model, the block-fading AWGN channel. This model is suitable to represent wireless transmission conditions with respect to time-variance and resource limitations. Suitable error control schemes based on punctured high-memory conventional codes and appropriate decoding methods are applied. Rate-control based on rate-distortion criteria allows selecting optimized source and channel coding rates for both video-coding schemes. In addition, the exploitation of network feedback - typically available in conversational services - is employed using NEWPRED techniques. A fair comparison of the systems shows that the standard-compliant non-scalable system outperforms the advanced scalable systems in terms of average decoded PSNR. The benefits of the scalable system are discussed and also presented via experimental results.

Video transmission in wireless environments is a challenging task calling for high-compression efficiency as well as a network friendly design. Both have been major goals of the H.264/AVC standardization effort addressing "conversational" (i.e., video telephony) and "nonconversational" (i.e., storage, broadcast, or streaming) applications. The video compression performance of the H.264/AVC video coding layer typically provides a significant improvement. The network-friendly design goal of H.264/AVC is addressed via the network abstraction layer that has been developed to transport the coded video data over any existing and future networks including wireless systems. The main objective of this paper is to provide an overview over the tools which are likely to be used in wireless environments and discusses the most challenging application, wireless conversational services in greater detail. Appropriate justifications for the application of different tools based on experimental results are presented.

Summary form only given. Source-controlled resource allocation is investigated for orthogonal transmission of heterogenous sources over a multiuser system. Different scenarios are considered where the system resources rate, bandwidth, and power are shared among all users in such a way that the average distortion over all users is minimized. This may prove beneficial to understand the trade-offs in mobile communication broadcast scenarios where common system resources are allocated depending on the instantaneous application needs. In a mobile communications environment, the channels to the users affect the transmitted signal differently so that rates cannot be traded for each other but bandwidth and transmission time are fairly easy tradable.

We investigate different allocation problems in mobile ad hoc networks. The objective is to minimize the power consumption and end-to-end distortion when transmitting images. Therefore, we use progressive image coding in combination with unequal erasure protection. We investigate two different scenarios, independent packet loss events and path loss events. For both cases, the expected distortion for progressive image transmission without error protection as well as for unequal erasure protection has been derived. Also, the expected power consumption for a certain path allocation pattern is obtained. The construction of operational distortion-power curves is presented by optimizing the unequal error protection and extracting the convex hull of all possible path allocation patterns. The profit of applying UEP to the progressively coded image bitstream in a multipath environment is shown. The presented error protection and rate allocation can be used as a framework for studying the allocation problem concerning the tradeoff between power and distortion for progressive image transmission in different multipath environments.

In this work we present a video transmission system for packet lossy channels based on the progressive texture video coding (PTVC) algorithm. A posteriori video rate control on the progressively coded messages is simplified significantly as for rate adaptation only the later part of this message can be truncated. This property is used by an appropriate unequal erasure protection (UXP) scheme which allows us - in combination with the progressively coded messages - to transmit very efficiently over packet erasure channels like the Internet. An appropriate rate allocation scheme based on dynamic programming is introduced. Additionally, a source-channel coding system exploiting network feedback is presented. Dependency on feedback delay is studied. Information between source encoder, channel encoder and network is exchanged and exploited in a cooperative way. The investigated systems show significant gains for different video sequences and different channel erasure rates when compared to sophisticated standard approaches.

The JVT/H.26L video coding scheme conceptually consists of a video coding layer (VCL) responsible mainly for coding efficiency and a network abstraction layer (NAL) that supports video specific transport features for a variety of networks. This paper describes the H.26L/JVT NAL in general, including the parameter set concept and the transport over packet-based and bit-stream oriented networks. Encapsulation of coded video data to RTP/UDP/IP transport is discussed, and applications of H.26L in fixed and wireless IP-based transmission environments are presented.

Transmission of hybrid coded video including motion compensation and spatial prediction over error-prone channels results in the well-known problem of spatio-temporal error propagation at the decoder. A widely accepted standard-compliant technique to enhance the quality of the decoded video significantly is the more frequent introduction of intra-coded macroblocks. However, intra-coded information generally requires more bit rate. Therefore, a careful selection of intra-updates in terms of rate and distortion is necessary. A flexible and robust rate-distortion optimization technique is presented to select coding mode and reference frame for each macroblock. The channel statistics are included in the optimization process. We derive a method to obtain an estimate of the decoder pixel distortion at the encoder. The presented techniques are verified within the new H.26L/JVT video coding standard based on common test conditions.

We present and compare several video transmission systems with various feedback and coding strategies for wireless real-time transmission. A channel coding system based on high-memory convolutional codes (CCs) is deployed which aims at delaying the first error within a progressively coded data frame as far as possible. A low-rate feedback channel is introduced to communicate the number of error-free received bits to the transmitter or to request additional redundancy. Thus, drift problems are avoided, since video encoder and decoder predict from identical reference frames. Several video communication systems based on forward error correction (FEC) and automatic repeat request (ARQ) methods are presented and compared. Experimental results are provided. The benefits of the proposed combination of unequal error protection (UEP) or ARQ systems with progressively coded sources and network feedback are highlighted.

We present a new resource allocation strategy that allows efficient mapping of variable-rate IP-streams onto shared cellular links, while minimizing the overall bandwidth consumption and still meeting the individual quality-of-service constraints. It is based on the general principle of class-based queuing, with several modifications to adapt it to the specific characteristics of wireless links. Compared to other simpler solutions that do not support any prioritization of data streams, it yields significant performance improvements in terms of guaranteed bandwidth per user/service and fairness among competing streams, as is shown for a GSM-GPRS system.

Simple but meaningful models for a mobile radio channel are introduced and a channel-coding system based on high-memory rate-compatible punctured convolutional codes with an appropriate sequential decoding algorithm, the far-end error decoder (FEED), are presented. In combination with puncturing, we devise a method for unequal error protection (UEP) and error localization within a progressively coded source message without any additional error detection code. The FEED-based channel-coding system does not aim to minimize the bit or word error probability, but to delay the first error within a data frame as far as possible. This channel-coding scheme and the FEED algorithm can be used efficiently with automatic repeat request (ARQ). We present different ARQ strategies. For all forward error-correction (FEC) schemes, bounds are specified which allow the estimation of the performance and appropriate rate allocation. We briefly discuss an efficient fine granular scalable video compression scheme, the progressive texture video codec (PTVC). The proposed scheme generates an embedded bit-stream for each frame and allows reference frames to be adjusted. These source and channel-coding algorithms are used to design several video communication systems based on FEC and ARQ methods. The resulting systems are presented and compared. Performance estimations based on bounding techniques and optimized rate-allocation algorithms are derived and applied. Experimental results show the improvement potential of the proposed systems compared to standard schemes. Video communication over very low bit-rate mobile channels with varying channel conditions is thus made possible.

We present two new decoding algorithms that estimate the output (path in the trellis) of a Markov source observed through a discrete memoryless channel with special application to convolutional codes. In a change of paradigm, these decoders do not exclusively aim at a low average error probability, but rather try to maximize the contiguously correct decoded subpath (where a subpath begins with the first symbol) - a concept termed far end error decoding. In addition to path estimation, these decoders provide a reliability about each subpath and, thus, enable us to localize the first decoding error without spending additional redundancy. This approach is motivated by the fact that in many applications it is much more important to deliver only error free data to the source decoder rather than to achieve a low error probability. The significant performance gain possible with this new approach is demonstrated for the example of SPIHT coded images.

We present a novel video coding approach which provides the functionality of fine granular bitstream scalability. The proposed progressive texture video coding (PTVC) scheme consists of two parts: the current ITU-T H.26L test model for coding macroblock mode and motion information only; a new technique for progressively coding all intra and inter texture information thus providing an embedded bitstream. It can still be decoded with moderate quality degradation in case of partial loss of the bitstream. It is therefore well suited for multicast or broadcast scenarios where time varying or different fixed bitrates are provided to the clients, as no transcoding or multiple coding process is required.

We present an efficient fine granular scalable video compression scheme which supports a fast bit rate adaptation independent of the encoder. The proposed scheme generates an embedded bitstream for each frame or, by appropriate multiplexing, for each group of picture. This rate-scalability is supported by an embedded bitstream which allows decoding at multiple rates, or to be more specific at virtually any rate. Drift removing techniques based on intra refresh or feedback mechanisms are presented. We show the potential of this video codec for variable bit rate channels as well as in combination with an unequal erasure protection scheme for error robust and efficient transmission over packet erasure channels. Possible enhancements of the presented coding schemes are discussed.

We present a source-channel coding system which is well suited for transmission over channels with varying channel states as common in mobile environments or broadcast scenarios. The system consists of a progressive source coder, a high-memory convolutional code (CC) with puncturing for unequal error protection (UEP) and a sequential decoding unit referred to as far end error decoder (FEED). In contrast to the conventional design goal of channel coding systems to minimize the bit error probability the presented system aims on maximizing the contiguously error-free part of the source message. Additionally, FEED allows one to localize the first error in the decoded message and only the virtually error free part is delivered to the source decoder. For image transmission gains up to 1.2 dB in average PSNR can be obtained compared to the best known system with the same channel decoder complexity. Finally, FEED allows a trade-off between receiver complexity and quality.

We present a method for modeling the distortion-rate characteristics of compressed video sequences. It is assumed that the source coders rely on motion-compensated prediction, and thus error propagation will contribute significantly to the degradation of the reconstructed video. It is shown that through the use of the modeled characteristics source and channel coding rate can be allocated across frames in a video sequence in order to minimize the effects of error propagation.

Multimedia transmission has to handle a variety of compressed and uncompressed source signals such as data, text, image, audio, and video. On wireless channels the error rates are high and joint source/channel coding and decoding methods are advantageous. Also, the system architecture has to adapt to the bad channel conditions. Several examples of a joint design are given. We especially advocate the use of rate-compatible punctured systematic recursive convolutional (RCPRSC) codes which are show to lead to a straightforward and versatile unequal error protection (UEP) design. In addition, the high-end receiver could use soft outputs and source-controlled channel decoding for even better performance.