Hrvoje Jenkac
Ludwig Maximilian University of Munich
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Publication
Featured researches published by Hrvoje Jenkac.
IEEE Transactions on Multimedia | 2004
Thomas Stockhammer; Hrvoje Jenkac; Gabriel Kuhn
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.
IEEE Transactions on Circuits and Systems for Video Technology | 2002
Thomas Stockhammer; Hrvoje Jenkac; Christian Weiss
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.
international conference on communications | 2006
Timo Mayer; Hrvoje Jenkac; Joachim Hagenauer
Interference cancellation is an important issue since the coverage and capacity in the uplink are mainly interference limited. The interference from the own cell can be mitigated by multiuser detection. However, there does not exist a powerful interference cancellation technique for the interference from neighboring cells, the intercell interference. This work proposes a technique for combating the intercell interference. Since base-stations in wireless systems are connected through a backbone network, we propose the exchange of information through the network between the base-stations, in order to help each other in the detection of the signals. We present several strategies how this could be performed. Especially, we propose an iterative way for reducing the intercell interference: The Turbo Base-Station Cooperation. Simulation results show remarkable gains, especially in the presence of a strong interferer.
wireless communications and networking conference | 2005
Hrvoje Jenkac; Günther Liebl; Thomas Stockhammer; Wen Xu
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.
IEEE Network | 2006
Hrvoje Jenkac; Thomas Stockhammer; Wen Xu
We discuss wireless broadcasting of multimedia streams within a framework that allows 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. Erasure-based decoding as well as soft decoding is discussed. Asynchronous data reception and full reliability are achieved at the same time. Depending on its receiving conditions, the receiver adapts its initial playout delay to guarantee high reliability of successful playout
vehicular technology conference | 2004
Hrvoje Jenkac; Günther Liebl; Thomas Stockhammer; Wen Xu
In this work, we present the multimedia broadcast multicast services (MBMS) framework for GSMEDGE radio access network (GERAN). Flexible outer Reed-Solomon coding is introduced in the GERAN MBMS environment. Performance estimations are presented. These performance estimations are used to compare the flexible RS coding scheme with simple repetition schemes. It is shown that at a target C/I of 7.5 dB and a maximum throughput of 40 kbit/s can be supported. Applying the flexible Reed-Solomon coding we also show that, e.g. for a C/I of 15 dB, even more than 90 kbit/s can be supported by MBMS.
international conference on multimedia and expo | 2005
Hrvoje Jenkac; Thomas Stockhammer
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
communications and mobile computing | 2007
Thomas Stockhammer; Tiago Gasiba; Wissam Abdel Samad; Thomas Schierl; Hrvoje Jenkac; Thomas Wiegand; Wen Xu
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.
transactions on emerging telecommunications technologies | 2006
Hrvoje Jenkac; Joachim Hagenauer; Timo Mayer
Reliable wireless broadcast with asynchronous data access based on fountain coding is investigated. We review the traditional problem formalisation for fountain codes operating on erasure channels, and we generalise the problem formalisation to more general channels. We introduce a novel type of rateless codes based on the Turbo Principle: the Turbo-Fountain. The Turbo-Fountain is able to consider soft information from the channel in the decoding process. Two realisations for the Turbo-Fountain are introduced. We show simulation results for the Turbo-Fountain realisations on the AWGN and on fading channels. Additionally, we compare the achievable Turbo-Fountain performance with traditional fountain codes designed for the erasure channel, both on the AWGN and on fading channels, considering an appropriate erasure declaration. It is shown that the Turbo-Fountain provides significant performance gains, due to exploitation of soft information, and approaches capacity. Moreover, we investigate the Turbo-Fountain within the framework of joint source-fountain coding. We show that the Turbo-Fountain provides the additional ability of lossless source compression and outperforms a system with separate source coding followed by fountain coding.
international conference on networking | 2005
Giinther Liebl; Hrvoje Jenkac; Thomas Stockhammer; Christian Buchner
In this paper we revisit strategies for joint radio link buffer management and scheduling for wireless video streaming. Based on previous work [1], we search for an optimal combination of scheduler and drop strategy for different end–to–end streaming options. We will show that a performance gain vs. the two best drop strategies in [1], ie drop the HOL packet or drop the lowest priority packet starting from HOL, is possible: Provided that basic side-information on the video stream structure is available, a more sophisticated strategy removes packets from an HOL group of packets such that the temporal dependencies usually present in video streams are not violated. This advanced buffer management scheme yields significant improvements for almost all investigated scheduling algorithms and streaming options. In addition, we will demonstrate the importance of fairness among users when selecting a suitable scheduler.