We propose a new framework for multiuser detection in fast fading channels that are encountered in many mobile communication scenarios. 1 Existing multiuser RAKE receivers, developed to combat multipath fading and multiuser interference in slow fading, suffer substantial degradation in performance under fast fading due to errors in channel state estimation. The detectors proposed in this paper employ a novel receiver structure based on time-frequency processing that is dictated by a canonical representation of the wide-sense stationary uncorrelated scatterer channel model. The workhorse of the framework is a time-frequency generalization of the RAKE receiver that exploits joint multipath-Doppler diversity. Analytical and simulated results based on realistic fast fading assumptions demonstrate that the proposed multiuser detectors promise substantially improved performance compared to existing systems due to the inherently higher level of diversity afforded by multipath-Dop...

Recently it was proposed to adapt several transmission methods, including modulation, power control, channel coding and antenna diversity to rapidly time variant fading channel conditions. Prediction of the channel coefficients several tens-to-hundreds of symbols ahead is essential to realize these methods in practice. We describe a novel adaptive long range fading channel prediction algorithm (LRP) and its utilization with adaptive transmission methods. This channel prediction algorithm computes the linear Minimum Mean Squared Error (MMSE) estimates of future fading coefficients based on past observations. This algorithm can forecast fading signals far into the future due to its significant memory span, achieved by using a sufficiently low sampling rate for a given fixed filter size. The LRP is validated for standard stationary fading models, and tested with measured data and with data produced by our novel realistic physical channel model. This model accounts for the variation of the amplitude, frequency and phase of each reflected component of the fading signal. Both numerical and simulation results show that long range prediction makes adaptive transmission techniques feasible for mobile radio channels.

Joint time-variant channel estimation and multiuser detection are key building-blocks for wireless broadband communication for mobile users at vehicular speed. We propose an iterative receiver for a multi-carrier (MC) code division multiple access (CDMA) system in the uplink. Multi-user detection is implemented through iterative parallel interference cancelation and conditional linear minimum mean square error (MMSE) filtering. MC-CDMA is based on orthogonal frequency division multiplexing (OFDM), thus time-variant channel estimation can be performed for every subcarrier individually. The variation of a subcarrier over the duration of a data block is upper bounded by the maximum Doppler bandwidth which is determined by the maximum velocity of the users. We exploit results from the theory of time-concentrated and bandlimited sequences and apply a Slepian basis expansion for time-variant subcarrier estimation. This approach enables time-variant channel estimation without complete knowledge of the second-order statistics of the fading process. The square bias of the Slepian basis expansion is one order of magnitude smaller compared to the Fourier basis expansion. The square bias of the basis expansion is the determining factor for the performance of the iterative joint channel estimation and data detection. We present an iterative linear MMSE estimation algorithm for the basis expansion coefficients in a multi-user system. The consistent performance of the iterative receiver using the Slepian basis expansion is validated by simulations for a wide range of velocities.

Abstract—In video communications over error-prone channels, compressed video streams are extremely sensitive to bit errors. Often random and burst bit errors impede correct decoding of parts of a received bitstream. Video decoders normally utilize error concealment techniques to repair a damaged decoded frame, but the effectiveness of these error concealment schemes relies heavily on correctly locating errors in the bitstream. In this paper, we propose a fragile watermark-based error detection and localization scheme called “force even watermarking (FEW)”. A fragile watermark is forced onto quantized DCT coefficients at the encoder. If at the decoder side the watermark is no longer intact, errors exist in the bitstream associated with a particular macro-block (MB). Thanks to the watermark, bitstream errors can accurately be located at MB level, which facilitates proper error concealment. This paper describes the algorithm, model and analysis of the watermarking procedure. Our simulation results show that compared to the syntax-based error detection schemes, the proposed FEW scheme significantly improves the error detection capabilities of the video decoder, while the peak signal-to-noise ratio loss and additional computational costs due to watermark embedding and extraction are small. Index Terms—Error detection, error resilience, forced even watermark (FEW), fragile watermark, video coding, video communications, wireless communications. I.

. We propose and analyze the performance of an algorithm for integrated power control and adaptive modulation/coding to achieve a specified range of packet error rate for real-time applications in broadband wireless packet-switched networks. The algorithm applies the Kalman-filter method [1] for power control, and adapts packet transmission to an appropriate modulation level, according to the packet error requirement, and the radio and interference conditions. A new criterion for maintaining stable transmission power is derived. Based on the criterion, the proposed technique performs the link adaptation and adjusts transmission power to achieve the specified packet error rate. The effectiveness of the proposed method is demonstrated by several numerical examples. 1 INTRODUCTION Customers' demand for broadband network services has been growing significantly as telecommuting and Internet access become increasingly popular. In the very near future, broadband services are also exp...

This paper presents a discussion of several middleware design issues related to the support of real-time multimedia communications over wireless personal communication services (PCS) networks. Specific interests are given to error recovery and synchronization mechanisms. A hybrid automatic repeat request (ARQ) scheme is employed for error control in the proposed system because it can efficiently adapt to nonstationary wireless channels and yield high throughput and reliability. In particular, delay and delay jitter control related to retransmissions in the error control module are addressed. An adaptive source rate control mechanism is used to handle the fluctuation of the effective channel data rate due to retransmissions. An adaptive synchronization scheme is developed to compensate for long-term delay variation caused by large-scale fading so that synchronization is preserved and end-to-end delay is kept low. Simulation results from the performance evaluation of the system are presented.

This paper investigates the performance of a turbo coded system transmitting over correlated flat fading channels with channel interleaving and both perfect and esti- mated channel state information. We show that channel in- terleaving is necessary on a correlated fading channel, and that performance degrades as the product of fading band- width and symbol duration (BTs) decreases. We consider both Rayleigh and Rician fading, and compare the per- formance of Log-Maximum A PosterJori (Log-MAP) and Soft Output Viterbi Algorithm (SOVA) based decoding al- gorithms. It is shown that for fading channels, the performance of the Log-MAP algorithm is considerably superior to that of the SOVA algorithm. A simple method for estimating the fading amplitudes and noise variance is proposed, and the impact of its use is investigated. It is shown that estimating the fades degrades performance only slightly, and estimating the noise variance does not noticeably affect performance.

The global communications system today (the telephone system yesterday) is considered as the largest man-made system all categories. While the demand for increased bandwidth in such systems increases, an increased interest in utilizing the available resources efficiently can be observed. Here, the subset of wireless cellular communications systems will be in focus and methods for transmitter power control.

Providing quality of service (QoS) guarantees is an important objective in the design of the next-generation wireless networks. In this dissertation, we address the QoS provisioning problem from both the network and the end system perspectives. In the first

Abstract. With the scalability and flexibility of the MPEG-4 and the emergence of the broadband wireless network, wireless multimedia services are foreseen to become deployed in the near future. Transporting MPEG-4 video over the broadband wireless network is expected to be an important component of many emerging multimedia applications. One of the critical issues for multimedia applications is to ensure that the quality-of-service (QoS) requirement to be maintained at an acceptable level. This is further challenged in that such a service guarantee must be achieved under unreliable and time-varying wireless channels. In this paper we study the link level performance of MPEG-4 video transmission over the uplink of an unreliable wireless channel. We introduce the discrete time batch Markovian arrival process (DBMAP) with two types of arrivals to model the MPEG-4 video source, which takes into account the inherent nature of the adaptiveness of the video traffic. We prove that in a hidden Markov modeled (HMM) wireless channel with probabilistic transmission, the service time for an arbitrary radio link control (RLC) burst follows phase type (PH-type) distribution. We show that the link level performance of a wireless video transmission system can be modeled by a DBMAP/PH/1 priority queue, and present computation algorithm and numerical results for the queueing model. Extensive simulations are carried out on the queueing behavior of the video transmission buffer, as well as on the packet level error behavior of the video data. The results demonstrate that video quality can be substantially improved by preserving the high priority video data during the transmission. Keywords: DBMAP with marked transitions, HMM channel, PH-type distribution, DBMAP/PH/1 priority queue 1.