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The closed-loop transmit diversity technique is used to improve the performance of the downlink channel in multiple-input multiple-output (MIMO) communication systems. The wideband code-division multiple-access standard endorsed by the Third Generation Partnership Project (3GPP) adopts a downlink closed-loop scheme based on partial channel state information (CSI) known as mode 1. The information is fed back from the mobile unit to the base station through a low-rate uncoded feedback bit stream. In this paper, several reconstruction techniques are introduced to improve the performance of mode 1 of 3GPP in the presence of feedback error by taking advantage of the redundancy available in the bit stream of the CSI. We propose a number of algorithms for the reconstruction of beamforming weights at the base station with the constraint of a constant transmit power. The performance is examined within a simulated 3GPP framework. It is demonstrated that the proposed algorithms have substantial gain over the conventional method for low, moderate, and high mobile speeds. It is also shown that the proposed algorithms can substitute the antenna weight verification process done at the mobile unit in many cases. The proposed approach is applicable to other feedback schemes as well. [ABSTRACT FROM AUTHOR]

In this paper, a method is proposed to construct spectrally efficient unitary space-time codes for high-rate differential communications over multiple-antenna channels. Unlike most of the known methods which are designed to maximize the diversity product (the minimum determinant distance), our objective is to increase the spectral efficiency. The simulation results indicate that for high spectral efficiency and for more than one receive antenna, the new method significantly outperforms the existing alternatives. In the special case of two transmit antennas, which is the main focus of this paper, the relation between the proposed code and the Alamouti scheme helps us to provide an efficient maximum-likelihood (ML) decoding algorithm. Also, we demonstrate that similar ideas can be applied to designing codes for more than two transmit antennas. As an example, we present a construction for 4-by-4 unitary constellations which has a good performance, compared with the other known codes. [ABSTRACT FROM AUTHOR]

A lattice-based scheme for the single-frame and the double-frame quantization of the speech line spectral frequency parameters is proposed. The lattice structure provides a low-complexity vector quantization framework, which is implemented using a trellis structure. In the single-frame scheme, the intraframe dependencies are exploited using a linear predictor. In the double-frame scheme, the parameters of two consecutive frames are jointly quantized and hence the interframe dependencies are also exploited. A switched scheme is also considered in which, lattice-based double-frame and single-frame quantization is performed for each two frame and the one which results in a lower distortion is chosen. Comparisons to the Split-VQ [13], the Multi-Stage VQ [12], the Trellis Coded Quantization [15], the interframe Block-Based Trellis Quantizer [17], and the interframe scheme used in IS-641 EFRC [26] and the GSM AMR codec [28] are provided. These results demonstrate the effectiveness of the proposed lattice-based quantization schemes, while maintaining a very low complexity. Finally, the issue of the robustness to channel errors is investigated. [ABSTRACT FROM AUTHOR]

This paper describes a new approach to fixed-rate entropy-constrained vector quantization (FEVQ) for stationary memoryless sources where the structure of codewords are derived from a variable-length scalar quantizer. We formulate the quantization search operation as a zero-one integer-optimization problem, and show that the resulting integer program can be closely approximated by solving a simple linear program. The result is a Lagrange formulation which adjoins the constraint on the entropy (codeword length) to the distortion. Unlike the previously known methods with a fixed Lagrange multiplier, we use an iterative algorithm to optimize the underlying objective function, while updating the Lagrange multiplier until the constraint on the overall rate is satisfied. The key feature of the new method is the substantial reduction in the number of iterations in comparison with previous related methods. In order to achieve some packing gain, we combine the process of trellis-coded quantization with that of FEVQ. This results in an iterative application of the Viterbi algorithm on the underlying trellis for selecting the Lagrange multiplier. Numerical results are presented which demonstrate substantial improvement in comparison with the alternative methods reported in the literature. [ABSTRACT FROM AUTHOR]

In this paper, the problem of reducing the peak-to-average-power ratio (PAPR) in an orthogonal frequency-division multiplexing system is considered. We design a cubic constellation, called the Hadamard constellation, whose boundary is along the bases defined by the Hadamard matrix in the transform domain. Then, we further reduce the PAPR by applying the selective-map- ping technique. The encoding method, following the method introduced in the work of Kwok, is derived from a decomposition known as the Smith normal form. This new technique offers a PAPR that is significantly lower than those of the best-known techniques without any loss in terms of energy and/or spectral efficiency, and without any side information being transmitted. Moreover, it has a low computational complexity. [ABSTRACT FROM AUTHOR]

A technique is presented to reduce the multiple-access interference in the forward link of a direct-sequence code-division multiple-access system. For each symbol period, an energy-constrained nonlinear transformation is applied at the transmitter output to minimize the mean-squared error at the receiver, subject to a constraint on the peak transmitted energy. The proposed algorithm can be implemented with existing optimization techniques that solve the quadratic trust-region problem. It is shown that in the presence of forward error-correction codes, the proposed method results in a significant gain over earlier known techniques at the cost of a modest increase in computational complexity at the base station. Another advantage of the proposed precoder is that it caps the peak energy (while earlier methods cap the average energy), requiring less sophisticated power amplifiers. [ABSTRACT FROM AUTHOR]

In this paper, we address the application of adaptive modulation and coding (AMC) for 3rd-generation (3G) wireless systems. We propose a new method for selecting the appropriate modulation and coding schemes (MCS) according to the estimated channel condition. In this method, we take a statistical decision making approach to maximize the average throughput while maintaining an acceptable frame error rate (FER). We use a first order finite state Markov model to approximate the time variations of the average channel signal to noise ratio (SNR) in subsequent frames. The MCS is selected in each state of this Markov model (among the choices proposed in the 3G standards proposals) to maximize the statistical average of the throughput in that state. Using this decision making approach, we also propose a simplified Markov model with fewer parameters, which is suitable in systems where changes in the fading characteristics need to be accounted for in an adaptive fashion. Numerical results are presented showing that both of our models substantially outperform the conventional techniques that use a memoryless threshold based decision making. [ABSTRACT FROM AUTHOR]

In this paper, deliberate level clipping and turbo coding are combined to achieve an orthogonal frequency division multiplexing (OFDM) transmission system with a low peak-to-average power ratio (PAR) and a good performance. Using the linear approximation technique based on the minimum mean square error (MMSE) criterion, we first modify the metric computation for the turbo decoding in order to consider the distortion effects of the nonlinearity, caused by the Cartesian clipper. Also, this paper introduces a modified turbo decoder which simultaneously performs the data estimation and signal reconstruction. In other words, the turbo decoder iteratively recovers the clipped signal by using the estimated data, and then improves the data estimation by using the newly recovered signal. Numerical results are presented showing an improvement in the performance of the OFDM transmission system over the nonlinear channel, an increase in the efficiency of the high power amplifier (HPA), and/or an expansion of the transmitter coverage area. [ABSTRACT FROM AUTHOR]

A new method of combined source-channel coding for the scalar quantization of a discrete memoryless source is presented, which takes advantage of the reliability information produced by a soft-output channel decoder. Numerical results are presented fot a memoryless Gaussian source in conjunction with turbo code showing up to 1-dB improvement in the end-to-end distortion with respect to a traditional channel optimized scalar quantizer. The results include a Gaussian source designed using closed-form expression without the need for a training sequence, as well as wage pixels using a training sequence. Furthermore, certain issues related to the effect of the channel mismatch and spectral efficiency of the system are studied. It is shown that the increase in distortion due to a channel mismatch can be substantially reduced by using an adaptive receiver. [ABSTRACT FROM AUTHOR]

Performance evaluation of maximum-likelihood (ML) soft-decision-decoded binary block codes is usually carried out using bounding techniques. Many tight upper bounds on the error probability of binary codes are based on the so-called Gallager's first bounding technique (GFBT). The tangential sphere bound (TSB) of Poltyrev which has been believed for many years to offer the tightest bound developed for binary block codes is an example. Within the framework of the TSB and GFBT, we apply a new method referred to as the "added-hyper-plane" (AHP) technique, to the decomposition of the error probability. This results in a bound developed upon the application of two stages of the GFBT with two different Gallager regions culminating in a tightened upper bound beyond the TSB. The proposed bound is simple and only requires the spectrum of the binary code. [ABSTRACT FROM AUTHOR]

The error probability of maximum-likelihood (ML) soft-decision decoded binary block codes rarely accepts nice closed forms. In addition, for long codes, ML decoding becomes prohibitively complex. Nevertheless,bounds on the performance of ML decoded systems provide insight into the effect of system a parameters on the overall system performance as well as a measure of goodness of the suboptimum decoding methods used in practice. Using the so-called Gallager's first bounding technique (involving a so-called Gallager region) and within the framework of tangential sphere bound (TSB) of Poltyrev, we develop a general bound referred to as the generalized TSB (GTSB). The Gallager region is chosen to be a general hyper-surface of revolution (HSR) which is optimized to tighten the bound. The search for the optimal Gallager region is a classical problem dating back to Glllager's thesis in the early 1960s For the random coding case, Gallager provided the optimal solution in a closed form while for the none random case the problem has been an active area of research in information theory for many years. We prove that for a sphere code, the optimal HSR within the proposed GTSB is a hyper-cone. This will climax to the TSB of Poltyrev, one of the tightest bounds ever, developed for binary block codes, and therefore terminates the search for a better Gallager region in the groundwork of the GTSB. [ABSTRACT FROM AUTHOR]

In this paper, we consider the problem of decoding a predictively encoded signal over a noisy channel when there is a residual redundancy (captured by a γ-order Markov model) in the sequence of transmitted data. Our objective is to minimize the mean-squared error (MSE) in the reconstruction of the original signal (input to the predictive source coder). The problem is formulated and solved through minimum mean-squared error (MMSE) decoding of a sequence of samples over a memoryless noisy channel. The related previous works include several maximum a posteriori (MAP) and MMSE-based decoders. The MAP-based approaches are suboptimal when the performance criterion is the MSE. On the other hand, the previously known MMSE-based approaches are suboptimal, since they are designed to efficiently reconstruct the data samples received (the prediction residues) rather than the original signal. The proposed scheme is set up by modeling the source-coder-produced symbols and their redundancy with a trellis structure. Methods are presented to optimize the solutions in terms of complexity. Numerical results and comparisons are provided, which demonstrate the effectiveness of the proposed techniques. [ABSTRACT FROM AUTHOR]

An analytical method for approximate performance evaluation of binary linear block codes using an additive white Gaussian noise channel model with binary phase-shift keying modulation is presented. We focus on the probability density function of the bit log-likelihood ratio (LLR), which is expressed in terms of the Gram-Charlier series expansion. This expansion requires knowledge of the statistical moments of the bit LLR. We introduce an analytical method for calculating these moments. This is based on some recursive calculations involving certain weight enumerating functions of the code. It is proved that the approximation can be as accurate as desired, if we use enough terms in the Gram-Charlier series expansion. Numerical results are provided for some examples, which demonstrate close agreement with simulation results. [ABSTRACT FROM AUTHOR]

Proposes a block-based trellis quantization (BTQ) scheme for the quantization of the line spectral frequencies (LSF) in speech coding applications. Properties of the LSF parameters; Overview of the BTQ design algorithm; Intraframe coding schemes of LSF parameters.

Presents a method for maximum likelihood decoding of the quadratic residue code. Basis of the method on projecting the code onto a subcode with an acyclic Tanner graph; Development of a two level coset decoding which can be considered a systematic generalization of the Wagner rule; Determination of the highest minimum distance of a binary code having a Pless-construction.

Proposes a number of different iterative decoding methods for the multiuser interference cancelation in a code-division multiple-access (CDMA) system where turbo-codes are utilized for forward error correction. Direct-sequence CDMA system model; Iterative multiuser CDMA receiver; Numerical results; Conclusion.

Examines the complexity of decoding lattices used in signal constellations by using the Korkin-Zolotarev reduced basis. Main applications of lattices in communications; Explanation on the concept of coding gain and ESM lattices; Information about the basis reduction; Discussion of the Korkin-Zolotarev (K-Z) reduced basis.

Delay, guaranteeing causality, is inevitable in cooperative communication systems. Traditionally, delay granularity has been limited to one symbol; however, channel delay is in fact governed by channel memory and can be shorter. For example, the delay requirement in orthogonal frequency-division multiplexing, captured in the cyclic prefix, is typically much shorter than the symbol itself. This perspective is used to study the two-user Gaussian interference channel with full-duplex transmitters. By superimposing the signal from the other node onto its own signal, each transmitter cancels the interference at its receiver. Among other results, it is proved that under a mild condition, the maximum multiplexing gain of this channel is in fact two, rather than the limit of one, previously shown under the traditional constraint of causal delay. Further, the optimal power allocation among orthogonal sub-carriers, which maximizes the achievable sum-rate, is shown to be a generalization of the well-known water filling. Simulation results are included to demonstrate the improvement in the achievable sum-rate when full-duplex transmitters are used. [ABSTRACT FROM AUTHOR]

It is known that the achievable data rate per user can be increased when relays are deployed in wireless networks. However, the drawback of this solution is that some of the network's resources should be allocated to the relays. In this paper, we consider a two-tier network in which all users send or receive data in two hops. By applying vector quantization, we compute the relays' locations to improve network's average transmission rate. These locations are also computed analytically when the number of relays is less than six. Having determined the relays' locations, the network's average transmission rate is evaluated. Subsequently, we define the "neutrality-surface" such that the performance of any relay network operating below this surface is inferior to that of the same network without relays. Finally, we study the relative relaying gain for different network configurations. [ABSTRACT FROM PUBLISHER]