Ericsson (Germany)

This paper describes a database designed to evaluate the performance of speech recognition algorithms in noisy conditions. The database may either be used to measure frontend feature extraction algorithms, using a defined HMM recognition back-end, or complete recognition systems. The source speech for this database is the TIdigits, consisting of connected digits task spoken by American English talkers (downsampled to 8kHz). A selection of 8 different real-world noises have been added to the speech over a range of signal to noise ratios with controlled filtering of the speech and noise. The framework was prepared as a contribution to the ETSI STQ-AURORA DSR Working Group[1]. Aurora is developing standards for Distributed Speech Recognition (DSR) where the speech analysis is done in the telecommunication terminal and the recognition at a central location in the telecom network. The framework is currently being used to evaluate alternative proposals for front-end feature extraction. The database has been made publicly available through ELRA so that other speech researchers to evaluate and compare the performance of noise robust algorithms. Recognition results will be presented for the first standard DSR feature extraction scheme based on a cepstral analysis. 1.

Multimedia watermarking technology has evolved very quickly during the last few years. A digital watermark is information that is imperceptibly and robustly embedded in the host data such that it cannot be removed. A watermark typically contains information about the origin, status, or recipient of the host data. In this tutorial paper, the requirements and applications for watermarking are reviewed. Applications include copyright protection, data monitoring, and data tracking. The basic concepts of watermarking systems are outlined and illustrated with proposed watermarking methods for images, video, audio, text documents, and other media. Robustness and security aspects are discussed in detail. Finally, a few remarks are made about the state of the art and possible future developments in watermarking technology.

This paper deals with 2/sup l/-ary transmission using multilevel coding (MLC) and multistage decoding (MSD). The known result that MLC and MSD suffice to approach capacity if the rates at each level are appropriately chosen is reviewed. Using multiuser information theory, it is shown that there is a large space of rate combinations such that MLC and full maximum-likelihood decoding (MLD) can approach capacity. It is noted that multilevel codes designed according to the traditional balanced distance rule tend to fall in the latter category and, therefore, require the huge complexity of MLD. The capacity rule, the balanced distances rules, and two other rules based on the random coding exponent and cutoff rate are compared and contrasted for practical design. Simulation results using multilevel binary turbo codes show that capacity can in fact be closely approached at high bandwidth efficiencies. Moreover, topics relevant in practical applications such as signal set labeling, dimensionality of the constituent constellation, and hard-decision decoding are emphasized. Bit interleaved coded modulation, proposed by Caire et al. (see ibid., vol.44, p.927-46, 1998), is reviewed in the context of MLC. Finally, the combination of signal shaping and coding is discussed. Significant shaping gains are achievable in practice only if these design rules are taken into account.

Work has started in the 3GPP to define a long-term evolution for 3G, sometimes referred to as super-3G, which will stretch the performance of 3G technology, thereby meeting user expectations in a 10-year perspective and beyond. The fundamental targets of this evolution - to further reduce user and operator costs and to improve service provisioning - will be met through improved coverage and system capacity as well as increased data rates and reduced latency. This article presents promising technologies to fulfil these targets, including OFDM, multi-antenna solutions, evolved QoS and link layer concepts, and an evolved architecture. Furthermore, the results of a performance evaluation are presented, indicating that the requirements can indeed be reached using the proposed technologies.

We propose an enhancement to TCP's error recovery scheme, which we call the Eifel algorithm . It eliminates the retransmission ambiguity, thereby solving the problems caused by spurious timeouts and spurious fast retransmits. It can be incrementally deployed as it is backwards compatible and does not change TCP's congestion control semantics. In environments where spurious retransmissions occur frequently, the algorithm can improve the end-to-end throughput by several tens of percent. An exact quantification is, however, highly dependent on the path characteristics over time. The Eifel algorithm finally makes TCP truly wireless-capable without the need for proxies between the end points. Another key novelty is that the Eifel algorithm provides for the implementation of a more optimistic retransmission timer because it reduces the penalty of a spurious timeout to a single (in the common case) spurious retransmission.

We survey the concept of assortativity, starting from its original definition by Newman in 2002. Degree assortativity is the most commonly used form of assortativity. Degree assortativity is extensively used in network science. Since degree assortativity alone is not sufficient as a graph analysis tool, assortativity is usually combined with other graph metrics. Much of the research on assortativity considers undirected, non-weighted networks. The research on assortativity needs to be extended to encompass also directed links and weighted links. In addition, the relation between assortativity and line graphs, complementary graphs and graph spectra needs further work, to incorporate directed graphs and weighted links. The present survey paper aims to summarize the work in this area and provides a new scope of research.

Demands for very high system capacity and end-user data rates of the order of 10 Gb/s can be met in localized environments by Ultra-Dense Networks (UDN), characterized as networks with very short inter-site distances capable of ensuring low interference levels during communications. UDNs are expected to operate in the millimeter-wave band, where wide bandwidth signals needed for such high data rates can be designed, and will rely on high-gain beamforming to mitigate path loss and ensure low interference. The dense deployment of infrastructure nodes will make traditional wire-based backhaul provisioning challenging. Wireless self-backhauling over multiple hops is proposed to enhance flexibility in deployment. A description of the architecture and a concept based on separation of mobility, radio resource coordination among multiple nodes, and data plane handling, as well as on integration with wide-area networks, is introduced. A simulation of a multi-node office environment is used to demonstrate the performance of wireless self-backhauling at various loads.

The fifth generation (5G) of cellular networks is starting to be defined to meet the wireless connectivity demands for 2020 and beyond. One area that is considered increasingly important is the capability to provide ultra-reliable and low-latency communication, to enable e.g., new mission-critical machine-type communication use cases. One such example with extremely demanding requirements is the industrial automation with a need for ultra-low latency with a high degree of determinism. In this paper, we discuss the feasibility, requirements and design challenges of an OFDM based 5G radio interface that is suitable for mission-critical MTC. The discussion is further accompanied with system-level performance evaluations that are carried out for a factory hall-wide automation scenario with two different floor layouts.

We find ourselves today often carrying numerous portable electronic devices, such as notebook computers, mobile phones, PDAs, digital cameras, and mp3/MD/DVD players, used to help and entertain us in our professional as well as private lives. For the most part, these devices are used separately, and their applications do not interact. Imagine, however, if they could interact directly and thus create a network where information may flow seamlessly between the devices-such a network of personal devices is often referred to as a personal area network, or PAN. Moreover, access to the Internet via a (public) wireless LAN access point and/or via a 3G UMTS mobile phone would enable the PAN to be constantly online. The strongest candidate to provide the cheap short-range radio links necessary to enable such networks is the Bluetooth wireless technology. Seen from a networking perspective, a PAN will be expected to have participants, both of its "own" devices and "guest" devices from other PANs, continuously moving in and out of its coverage. To cope with this volatile nature of the network, the concept of ad hoc networking may be applied to create robust and flexible connectivity. A major technical step is taken when the Bluetooth piconet network architecture, a strict star topology, is extended into a scatternet architecture, where piconets are interconnected. A consequence of creating scatternet-based PANs is that some nodes will form gateways between piconets, and these gateways must be capable of time sharing their presence In each piconet of which they are members. While the Bluetooth standard defines the gateway nodes, the actual mechanisms and algorithms that accomplish the interpiconet scheduling (IPS) are left rather open. Given the lack of research literature in the subject, an overall architecture for handling scheduling in a scatternet is presented. A family of feasible IPS algorithms, referred to as rendezvous point algorithms, is also introduced and discussed.

Enabling wireless Internet access is one of the upcoming challenges for mobile radio network operators. The General Packet Radio Service is the packet-switched extension of GSM and was developed to facilitate access to IP-based services compared to existing circuit-switched services provided by GSM. Besides an overview on the basic concept, network architecture, and protocols of GPRS, this article discusses the performance the end user perceives when retrieving information from the Web using this access technology. This discussion is based on results obtained from a protocol simulator comprising not only the radio interface protocols, but also the relevant Internet protocols as well as a characteristic application model. The results show that GPRS provides bandwidth-efficient support for bursty applications like Web access.

Future wireless systems are expected to be characterized by increasing convergence between networks and further development of reconfigurable radio systems. In parallel with this, demand for radio spectrum from these systems will increase, as users take advantage of high quality multimedia services. This article aims to investigate and review the possibilities for the dynamic allocation of spectrum to different radio networks operating in a composite reconfigurable wireless system. The article first looks into the current interest of regulators in this area, before describing some possible schemes to implement dynamic spectrum allocation and showing some example performance results. Following this, the technical requirements that a DSA system would have, in terms of reconfigurable system implementation, are discussed.

Recently, a lot of research effort has been spent on cross-layer system design. It has been shown that cross-layer mechanisms (i.e., policies) potentially provide significant performance gains for various systems. In this article we review several aspects of cross-layer system optimization regarding wireless OFDM systems. We discuss basic optimization models and present selected heuristic approaches realizing cross-layer policies by means of dynamic resource allocation. Two specific areas are treated separately: models and dynamic approaches for single transmitter/receiver pairs (i.e., a point-to-point communication scenario) as well as models and approaches for point-to-multipoint communication scenarios (e.g., the downlink of a wireless cell). This article provides basic knowledge in order to investigate future OFDM cross-layer-optimization issues

In this work, we consider a heterogeneous network consisting in several macro nodes and pico nodes. Our goal is to associate users, belonging to this network, to one of the nodes, while maximizing the sum rate of all users. We also want to analyze the load balancing achieved by this association. Therefore, we develop a new theoretical framework to study cell association for the downlink of multi-cell networks and derive an upper bound on the achievable sum rate. We propose a dynamic cell association heuristic, which achieves performance close to optimal. Finally, we verify our results through numerical evaluations and implement the proposed heuristic in an LTE simulator to demonstrate its viability.

At present, standards for broadband wireless multimedia communications in the 5 GHz band are being developed in Europe as well as in the USA and Japan. HIPERLAN/2 is an upcoming standard which is being specified by the ETSI Project BRAN and is intended to be finished in April 2000. The physical layers of the three systems will be well-harmonized whereas the upper layer protocols are different. In this paper, the concepts and parameters of the physical layer of HIPERLAN/2 are described. According to the harmonization with the American and the Japanese system similarities and differences are pointed out. Furthermore, exemplary performance results are included and discussed.

Abstract. Measuring rain rates over complex terrain is afflicted with large uncertainties, because rain gauges are influenced by orography and weather radars are mostly not able to look into mountain valleys. We apply a new method to estimate near surface rain rates exploiting attenuation data from commercial microwave links in the alpine region of Southern Germany. Received signal level (RSL) data are recorded minutely with small data loggers at the towers and then sent to a database server via GSM (Global System for Mobile Communications). Due to the large RSL fluctuations in periods without rain, the determination of attenuation caused by precipitation is not straightforward. To be able to continuously process the RSL data from July 2010 to October 2010, we introduce a new method to detect wet and dry periods using spectral time series analysis. Its performance and limitations are presented, showing that the mean detection error rates of wet and dry periods can be reduced to 10% for all five links. After, the wet/dry classification rain rates are derived from the RSL and compared to rain gauge and weather radar measurements. The resulting correlations differ for different links and reach values of R2 = 0.81 for the link-gauge comparison and R2 = 0.85 for the link-radar comparison.

LAA is a new operation mode of LTE in the unlicensed spectrum, which will be featured in LTE Release 13. Under LAA, licensed carriers will be aggregated with unlicensed carriers in order to opportunistically enhance downlink user throughput while still offering seamless mobility support. In order to coexist with WiFi, some of the new functionalities required of LAA LTE include a mechanism for channel sensing based on listen-before-talk, discontinuous transmission on a carrier with limited maximum transmission duration, and multicarrier transmission across multiple unlicensed channels. This article presents a detailed overview of the design agreements for LAA, the impact of unlicensed spectrum operation on the LTE physical layer architecture, and the scope of additional enhancements beyond LTE Release 13. A range of simulations for indoor and multicarrier scenarios show that fair coexistence between LAA and WiFi can be achieved, and that deployment of LAA can provide a boost in WiFi performance.

Models of visual attention provide a general approach to control the activities of active vision systems. We introduce a new model of attentional control that differs in important aspects from conventional ones. We divide the selection into two stages, which is more suitable for the system as well as explaining different phenomena found in natural visual attention, such as the dispute between early and late selection. The proposed model is especially designed for use in dynamic scenes. Our approach aims at modeling as much of a general active vision system as possible and designing clean interfaces for the integration of the remaining specific aspects needed in order to solve specific problems.

GPRS is the new packet-oriented data service for GSM. Soon it will be possible to take advantage of the features provided by GPRS for Internet access like fast connection set-up, volume-based charging and staying on-line for long periods. For non-real-time Internet applications TCP is the applied transport protocol. Often it was suspected that GPRS introduces severe performance degradations for TCP traffic since both apply their own ARQ mechanism. It is well known that layered protocol interactions can have very negative influences. This paper examines with simulations the performance of TCP over GPRS. The behaviour of the protocols is analysed under various conditions. It is shown that both TCP and GPRS are well harmonised.

MPEG has recently released a first public draft of the new Dynamic Adaptive Streaming over HTTP (DASH) specification. The new streaming technique is based on the 3GPP and partly also on the Open IPTV Forum (OIPF) specifications. In this paper, we give an overview about the new DASH specification with a special focus on Live streaming services. In order to re-use existing web content distribution schemes, the new streaming technique provides the live stream as a sequence of files, which are continuously downloaded by the streaming client. This way of streaming introduces new delay components into the system streaming. Live Streaming technology is often used for events like sport events to allow other users to virtually participate. It is generally preferred to minimize the end-to-end delay for live services. In the paper, we identify and analyze the different delay components of the new adaptive HTTP streaming technique and how they contribute to the end-to-end delay of live services. We also discuss the dependencies and system implications when minimizing the end to end delay. The evaluation principles are also applicable for other adaptive HTTP streaming formats.

There is a growing interest in over-the-top (OTT) dynamic adaptive streaming over Hypertext Transfer Protocol (HTTP) (DASH) services. In mobile DASH, a client controls the streaming rate and the base station in the mobile network decides on the resource allocation. Different from the majority of previous works that focus on client-based rate adaptation mechanisms, this paper investigates the mobile network potential for enhancing the user quality-of-experience (QoE) in multiuser OTT DASH. Specifically, we first present proactive and reactive QoE optimization approaches for adapting the adaptive HTTP video delivery in an long-term evolution network. We then show, using subjective experiments, that by taking a proactive role in determining the transmission and streaming rates, the network operator can provide a better video quality and a fairer QoE across the streaming users. Furthermore, we consider the playout buffer time of the clients and propose a novel playout buffer-dependent approach that determines for each client the streaming rate for future video segments according to its buffer time and the achievable QoE under current radio conditions. In addition, we show that by jointly solving for the streaming and transmission rates, the wireless network resources are more efficiently allocated among the users and substantial gains in the user perceived video quality can be achieved.