bluuebunny/crossref_metadata_2025_split huggingface dataset:
SMART: A rapid layer adaptation protocol for layered video multicast
Authors: Lee Jiwoong , Kim Jae‐Kyoon 
Published: September 2008

AbstractThis paper presents a new network layer traffic control protocol for rapid layer adaptation and prompt congestion recovery of a layered—video multicast session. The proposed protocol SMART, which stands for Scalable Multimedia Rapid Rate Control in Multicast Networks, defines a concept of dependency between multicast group addresses that are pertaining to the same layered—video session. The information on dependency is carried on an extended version of Internet Group Managenemt Protocol, which is called IGMP—DX (Dependency Extension). SMART router creates a database to manage the dependency information. With SMART, receivers can adapt their subscription level to the right level rapidly by the function of Fast Learning. Emergency Drop Request is sent by a host who detects intensive congestion, and the router blocks the associated traffic. In this manner network can recover from congestion rapidly while not impairing the minimal multicast service. SMART supports both IGMPv2 and IGMPv3, including sourcespecific multicast in Internet. It is shown by simulations that SMART provides a rapid rate control mechanism and good scalability with small amount of control signal.
Transport of MPEG—4 over IP/RTP
Authors: Basso Andrea , Varakliotis Socrates , Castagno Roberto , Lohan Florin 
Published: September 2008

AbstractIn this paper we review and discuss some of the current efforts and proposals in the standardization bodies IETF and MPEG for the delivery of MPEG—4 over IP. We address the issues of payload format and media control by describing two architectures that allow for MPEC—4 streaming. Finally, we describe some MPEG—4 over IP streaming experiments in which the RTP MPEG—4 payload format is used in conjunction with per—stream Forward Error Correction support, in presence of packet loss.
An end—to—end performance comparison methodology for ATM transport of CBR and VBR encoded digital video
Authors: Kurçeren Ragip , Modestino Jim W. 
Published: September 2008

AbstractCompressed digital video can be transported on ATM networks using either a constant bit rate (CBR) or a variable bit rate (VBR) network transport service. The use of VBR transport is generally considered to offer some potentially attractive advantages over more traditional CBR transport. These potential advantages include the expectation of improved reconstructed video quality, shorter end—to—end delays and improved statistical multiplexing performance. Unfortunately, most evaluations of the relative end—to—end performance of VBR versus CBR transport of digital video fail to adequately take into account the effects of lost cells in transporting multiple VBR streams which compete for limited network resources. These effects can be avoided with CBR transport using appropriate call admission control (CAC) procedures. Furthermore, these relative performance evaluations are generally not provided under comparable levels of network resource utilizations. Indeed, no methodology presendy exists for fairly assessing the relative end—to—end performance of CBR and VBR encoded video at the same level of resource utilization. In this paper, we develop and illustrate a methodology for comparing the relative performance, in terms of the corresponding end—to—end distortion, of CBR and VBR encoded digital video transport for fixed network resources and at identical multiplexing levels. The approach is an extension of a previously developed joint source—channel coding approach for VBR digital video transport on ATM networks. Our objective in this paper is to provide a common comparison framework for CBR and VBR video streams.
On improving CBR service playback buffer mechanism in WATM networks
Authors: Beben Andrzej , Burakowski Wojciech 
Published: September 2008

AbstractThe commonly recommended approach to meet the Constant Bit Rate (CBR) service requirements in ATM networks is to use the playback buffer mechanism implemented at the AAL layer [1],[2],[3]. This solution was originally proposed for the fibre optics ATM networks, where cell loss ratio is negligible (e. g. 10—9). However, for the wireless ATM network concept the designers have to take into account the fact that an essential part of sending cells can be simply lost due to transmission errors and these losses can also occur in bursts (despite applying error protection mechanisms). Moreover, the variation of cell transfer delay is expected to be significantly larger and this is mainly due to radio access protocol behaviour. As a consequence, the above leads to a need to modify the playback buffer mechanism in order to improve its functionality in the wireless ATM network environment. The solution proposed in this paper is based on some enhancements of the dummy cell insertion (instead of lost cells) algorithm and on the K times extension of AAL1 PDU numbering range, by assigning the same number to K consecutive cells. The justification of the approach is provided in terms of theoretical considerations and simulation results.
The application of modified lapped transform domain median filter to narrow—band interference excision in DSSS systems
Authors: Guangruihu Chongni Li 
Published: September 2008

AbstractA novel communication receiver which uses lapped transform (LT) incorporating modified median filter (MMF) algorithm is designed for narrow—band interference (NB1) excision. Comparing to traditional Fourier Transform, LT has longer basis vectors, less spectral leakage, thus better frequency resolution. The LT domain MMF algorithm takes full advantages of the direct sequence spread spectrum signal, as well as the characteristics of LT, performs the transform domain filtering twice. The first filtering locates the position of interference and mitigates most of them. The second filtering was performed in a small neighborhood of the located interference. So LT domain MMF algorithm can completely mitigate the interference without distorting the desired signal. Simulation results demonstrated the improved BER performance and increased robustness of our approach.
Feedforward joint phase and timing estimation for MSK—type signals
Authors: Morelli Michele , Vitetta Giorgio M. 
Published: September 2008

AbstractNovel non data—aided (NDA) algorithms are proposed for joint estimation of timing and carrier phase in MSK—type modulations. They are based on maximum likelihood methods and have a feedforward structure which is suitable to fully digital implementation. Performance with MSK and Gaussian MSK. (GMSK) is assessed by computer simulations and compared with that of other existing estimation schemes.
Exploiting the short—term and long—term channel properties in space and time: Eigenbeamforming concepts for the BS in WCDMA
Authors: Brunner Christopher , Utschick Wolfgang , Nossek Josef A. 
Published: September 2008

AbstractThe deployment of adaptive antennas at base stations considerably increases the spectral efficiency of wireless communication systems. To reduce the computational complexity and increase performance of space—time (ST) processing, processing may take place in reduced dimension, i. e., pre—filtering takes place which is related to linear estimation theory based on second order statistics. To this end, long—term and short—term channel estimates are integrated into specific Tx/Rx systems. In this article, we present a new ST rake structure for uplink reception in WCDMA which operates in reduced dimension. Accordingly, our approach combines short—term and long—term spatial and temporal channel properties using an eigenanalysis. By choosing dominant eigenbeams in time and space, the algorithm enhances interference suppression as well as spatial and temporal receive diversity. In contrast to previously introduced well—known receiver structures, the ST eigenrake inherently adapts to different propagation environments and achieves higher spectral efficiency than other receivers. This is illustrated by Monte—Carlo simulations.Then we extend the proposed concept to the downlink. The downlink eigenbeamformer improves closed—loop downlink diversity compared to other proposals in standardization (3GPP) which only exploit short—term channel properties. Even though the short—term feedback rate remains unchanged, additional antenna elements can be included to increase antenna and diversity gain. We also present a tracking solution to the downlink eigenbeamforming in WCDMA. To this end, we propose a distributed implementation of the eigenspace/—beam tracking at the mobile terminal and base station (BS), respectively. Moreover, the specific nature of the deployed tracking scheme offers an advantageous feedback signalling.
MIMO vector channel sounder measurement for smart antenna system evaluation
Authors: Thomä R. S. , Hampicke D. , Richter A. , Sommerkorn G. , Trautwein U. 
Published: September 2008

AbstractFor the simulation and design of smart antenna transmission principles in mobile radio, precise knowledge of the time—variant directional multipath structure in various radio environments is required. In this paper, a new real—time multiple—input—multiple—output (MIMO) vector radio channel sounder is described, which uses multiple antennas at the transmitter as well as at the receiver position. The proposed MIMO measurement principle can be effectively exploited to estimate the propagation direction at both ends of the wireless link simultaneously, and thus, dramatically enhance overall resolution of the multiple path parameters. Applying a proper antenna architecture and the multidimensional unitary ESPRIT algorithm, joint superresolution estimation of the direction of departure (DOD), time—delay of arrival (TDOA), Doppler shift, and direction of arrival (DOA) of the propagating waves becomes possible. The measured results can also be used directly for the simulation of combined transmit—receive diversity (MIMO) transmission principles and space—time (ST) adaptive receivers in a multi—user scenario. Results based on measurements in different locations are referenced, including a complicated indoor environment as is typical for industrial WLAN applications.
Partial imaginary precursor cancelling in DFE for BPSK and GMSK modulations
Authors: Raphaeli Dan , Stark Amit 
Published: September 2008

AbstractThis paper examines the effect of partial imaginary precursor cancelling on the performance of a system which uses a DFE and employs a one dimensional modulation. As opposed to QAM, BPSK or GMSK (after proper manipulations) require minimization of the mean square (MS) of the error real part only. Therefore, a lower MSE is achieved if the DFE coefficients are computed taking this fact into account, by trying to minimize as few imaginary precursors as possible. However, this introduces a delay by the imaginary part of the feedback filter (FBF) which is no longer causal.In this paper we investigate the influence of the length of imaginary precursor cancelling on system performance, and its effect on carrier phase tracking. The equations for the computation of the DFE coefficients for none or partially cancelled imaginary precursors are derived and performance on GMSK channels is presented.
Speech spectral quantizers for wideband speech coding
Authors: Guibé G. , How H. T. , Hanzo L. 
Published: September 2008

AbstractIn this treatise a rangeof Line Spectrum Frequency(LSF) Vector Quantization (VQ) schemes were studied comparatively, which were designed forwideband speech codecs.Both predictive arrangements and memoryless schemes were investigated. Specifically, both memoryless Split Vector Quantization (SVQ) and Classified Vector Quantization (CVQ) were studied. These techniques exhibit a low complexity and high channel error resilience, but require high bit rates for maintaining high speech quality. By contrast, Predictive Vector Quantizers (PVQ) offer an enhanced Spectral Distortion (SD) performance, although they are sensitive to channel error propagation. It is shown that the family of so—called Safety—Net Vector Quantization (SNVQ) schemes offers a good design compromise, providing an extension to memory—based PVQ, and thereby improving the performances both over noisy and noiseless channels.
A measurement based approach to TCP congestion control
Authors: Ruddle Alan , Allison Colin , Lindsay Peter 
Published: September 2008

AbstractCongestion is a major Internet problem. TCP has evolved in attemptting to deal with this and now employs congestion control algorithms that effectively limit the bandwidth available to any one connection. However, as TCP is not the only transport protocol used on the Internet, the growth of non‐TCP traffic has lead to the formulation of TCP‐Friendly formulae. It is recommended that all best effort traffic conform to these formulae, and they act as a benchmark against which the success of any congestion control algorithms can be judged. Unfortunately, key assumptions behind the formulation of TCP's algorithms — low bandwidth, and bulk data transfers — are often no longer valid. In this paper it is shown that the combination of increasing bandwidth and short web transfers means that a significant amount of TCP traffic fails to obtain the bandwidth implied by the TCP‐friendly formulae. An alternative, measurement‐based approach that predicts a fairer starting window size for a connection is presented and evaluated. Information about the characteristics of particular network paths is dynamically maintained and used to suggest a fairer starting window size for new connections. Minor modifications to TCP allow these suggestions to be used to set start‐up control variables, and thereby strengthen the negative correlation between the bandwidth used by a connection and the level of congestion on its network path. The measurement‐based approach is realised through the use of a Location Information Server (LIS). The LIS performs centralised passive monitoring of transport headers in order to derive network level path information. Location Information Packets (LIP) are used to communicate suggested start‐up variables to local hosts. The design, implementation and evaluation of an LIS, LIP and participating host's TCP software are presented.
A variable step‐size control of adaptation for acoustic echo cancellation filters
Authors: Pahor Vojko , Carini Alberto 
Published: September 2008

AbstractThis paper presents a low‐complexity variable step‐size control for acoustic echo cancellation filters. The proposed control method is based on two supplementary adaptive filters with low memory length. The first supplementary filter is used to control the double‐talk situations while the second supplementary filter is employed to detect variations of the echo‐path impulse response. The variable step‐size here derived is independent from die echo coupling loss and from the reverberation time and thus it is suitable for both hands‐free and handset terminals.
Frequency‐domain integration of acoustic echo cancellation and a generalized sidelobe canceller with improved robustness
Authors: Herbordt Wolfgang , Kellermann Walter 
Published: September 2008

AbstractFor hands‐free acoustic human/machine interfaces, as required, e.g., for automatic speech recognition, teleconferencing, and other multimedia services, microphone arrays using generalized sidelobe cancellers (GSCs) in conjunction with acoustic echo cancellation (AEC) can be efficiently applied for optimum communication. This contribution first devises a hew structure for combining AEC and GSC in the frequency domain. We show that computational complexity is reduced by more than a factor of ten compared to a time‐domain arrangement. Second, robustness issues of the GSC adaptation mechanism are addressed. We illustrate how robust adaptation is assured in the new structure. Third, we propose alternatives for the fixed beamformer part of the GSC, which allow to specify (a) the width of the target tracking region and (b) improve the suppression of low frequencies. GSC robustness constraints are included using linear optimization with configurable penalty functions.
Performance of microphone array geometries in a reverberant room
Authors: Ward Darren B. 
Published: September 2008

AbstractThe choice of sensor positions plays a critical role in determining the performance of a nearfield microphone array used for hands‐free speech acquisition. For a single source in a reverberant room, we propose a performance measure based on the expected reverberant power at the array output, and use the proposed measure to choose the microphone locations. One result of applying the proposed method is that it may be better to distribute the microphones throughout the room rather than placing them together in a single array (as conventional farfield array theory would dictate).
Hybrid quality measures for single‐channel speech enhancement algorithms
Authors: Dreiseitel Pia 
Published: September 2008

AbstractWe give an overview of speech quality measures and the most common noise reduction algorithms for the single‐channel approach including a description of their respective shortcomings and apply the quality measures to the special deficiencies of speech enhancement algorithms. This is important for comparing different approaches to noise reduction, and it suggests ways to improve a speech enhancement algorithm. Objective quality measures are then compared to the subjective quality evaluation of human listeners by means of extensive listening tests using the so‐called rank correlation. As a result, we propose a hybrid method for measuring the quality of speech enhancement algorithms.
Adaptive routing algorithms and buffer management in multihop metropolitan area networks
Authors: Giacomazzi Paolo , Melen Riccardo , Voltolini Mirko 
Published: September 2008

AbstractThe present study is devoted to multihop Metropolitan Area Network architectures characterized by some buffer capacity at the nodes and routing algorithm slightly more complex than those already present in the literature. A new adaptive routing algorithm and a new buffer management scheme are presented, and their performance are compared to that of the best algorithms known in the literature. Both the new proposals exhibit significant improvements in throughput and delay with respect to the older ones, for various MAN topologies, sizes and buffer dimensions. These results, which in some cases are very close to the theoretical optimum, are achieved with an implementation complexity well within the limits of present day technology.
Effects of phase noise spectral shape on the performance of dpsk systems for wireless applications
Authors: Corvaja Roberto , Pupolin Silvano 
Published: September 2008

AbstractPhase noise could be the major source of performance degradation in low cost low bit‐rate digital systems operating in the millimeter wavelength band. A general model considers the phase noise as the sum of several components, namely a white phase noise, a white frequency noise and a flicker frequency noise. All these components are taken into account in the evaluation of error probability in a DPSK receiver and are compared in terms of performance degradation.
Performance of a multicarrier ds/cdma system with successive interference cancellation
Authors: Huh Hoon , Kim Jin Young 
Published: September 2008

AbstractIn this paper, successive interference cancellation (SIC) scheme is proposed for a multicarrier DS/CDMA system and its performance is analyzed in a Rayleigh fading channel. The SIC scheme is employed to mitigate performance degradation due to multiple access interference (MAI) coupled with near‐far problem. Bit error probability of the proposed system is derived for maximal‐ratio and equal‐gain combining schemes of subcarrier signals, respectively. From numerical results, it is Shown that the proposed system achieves significant performance improvement over a conventional multicarrier DS/CDMA system. The proposed system can find application in multicarrier DS/CDMA systems for high‐rate data services.
Optical CDMA codes for use in a lightwave communication network with multiple data rates
Authors: Zhang Jian‐Guo 
Published: September 2008

AbstractThis paper presents a new kind of optical code‐division multiple‐access (CDMA) codes, called unequal‐length optical orthogonal codes (UL‐OOC's), for multirate data communications. The autocorrelation and cross‐correlation functions of UL‐OOC's are dependent only on each individual codeword, in spite of either an aperiodic or a periodic pulse‐sequence pattern at the input of optical CDMA decoders. This property can be thus used to support multirate data communications in a lightwave network, without any violation of the minimum correlation constraint (i.e., “1”) for incoherent optical processing. Theory and simple design of UL‐OOC's are presented. Applications of the proposed codes to multirate optical CDMA is also explained in detail.
Dwell time modeling for wireless communication systems and problems of the velocity distribution
Authors: Boche Holger , Jugl Enrico 
Published: September 2008

AbstractThe distribution of the dwell time, which represents a higher level of abstraction than the use of mobility models, is often required for the performance evaluation of mobile communication systems. This contribution investigates the problem of the dwell time modeling. Using the often employed model for a constant movement, it is shown that a suitable choice of the velocity distribution is very important. Thus, a condition for the probability density function (PDF) of the velocity is derived, which prevents an unbounded mean dwell time. Since an infinite mean dwell time causes some problems for me estimation of the mobility‐related signaling traffic, it is important that the mean dwell time is finite. Furthermore, some basic models for the PDF of the travel length of a subscriber within a radio cell are introduced and the relation wim realistic mobility patterns are shown.
A hidden pilot assisted demodulation technique for channel estimation in m‐qam transmissions
Authors: Mazzenga Franco 
Published: September 2008

AbstractMultipath effects represent the limiting factor for reliable wireless communications with high efficiency modulation schemes such as M‐QAM or M‐PSK. Pilot assisted modulation techniques have been proposed to overcome multipath fading effects and an equalizer operating on the basis of a received pilot is commonly deployed. The pilot sequence used to train the equalizer is usually time division multiplexed with the information sequence. This solution reduces the bandwidth efficiency and may become unusable when too frequent updating of the communications parameters is needed in the receiver. In this letter it is shown that bandwidth efficiency can be preserved at the expense of an increase in transmit power. The additional transmitted power is used to send a known pilot sequence hidden into the information sequence. The pilot is superimposed onto the information symbols to be transmitted before waveform modulation. Therefore each pilot symbol has the same period of the information symbol so that bandwidth spreading is avoided. The M‐QAM modulation format is considered but the technique can be extended to any digital communication system. The channel estimation problem is analyzed since the equalizer impulse response can be easily derived from the estimated channel impulse response. To prove the effectiveness of the proposed technique simulation results are provided.