This paper proposes a sequentially iterative equalizer based on Kalman filtering and smoothing (SIEKFS) for multiple-input multiple-output (MIMO) systems under frequency selective fading channels. In the proposed SIEKFS, an iteration consists of sequentially executed subiterations, and each subiteration performs equalization and detection procedures of the symbols transmitted from a specific transmit antenna. During this subiteration, all available observations for the transmission block are utilized in the equalization procedures. Furthermore, the entire soft estimate of the desired symbols to be detected does not participate in the equalization procedures of the desired symbols, i.e., the proposed SIEKFS performs input-by-input equalization procedures for a priori information nulling. Therefore, compared with the original iterative equalizer based on Kalman filtering and smoothing, which performs symbol-by-symbol equalization procedures, the proposed SIEKFS can also perform iterative equalization based on the Kalman framework and turbo principle, with a significant reduction in computation complexity. Simulation results verify that the proposed SIEKFS achieves suboptimum error performance as the size of the antenna configuration and the number of iterations increase.

Low-density parity-check (LDPC) codes are typically designed to avoid the length-4 cycles to ensure acceptable levels of performance. However, the turbo equalization, which relies on an interaction between an inner code such as an LDPC code and a soft-output Viterbi algorithm (SOVA) detector, exhibits a performance degradation due to the pseudo cycles. In this paper, we propose an interleaved modified array code (IMAC) that can reduce the number of pseudo cycles, hence, improving the gains from the iterative processing technique. The modification is made on the existing array-based LDPC codes named modified array codes (MAC) by introducing an additional interleaving matrix to the parity-check matrix. Simulation results on the perpendicular magnetic recording channels (PMRC) demonstrate that the IMAC outperforms both the MAC and the previously proposed random interleave array (RIA) codes for the partial-response targets under consideration. In addition, a subblock-based encoder design is proposed to reduce the encoding complexity of the IMAC and when compared with the RIA code, the IMAC exhibits a lower encoding complexity, and still maintains a comparable level of the decoding complexity.

This paper proposes an iterative peak power reduction method with adaptive intermediary over-sampling which uses the necessary minimum bandwidth according to iteration number for wireless OFDMA systems. The required bandwidth to each iteration number is evaluated by computer simulation, and over-sampling numbers in iterative processing are controlled by using the simulation results. The results show that the required bandwidth is 1.6, 2.0, and 2.7 times of the used signal bandwidth at the iteration number of 1, 2, and 3, respectively. The proposed adaptive over-sampling method can reduce its multiplication number by 13%.

Based on the expectation-maximization (EM) algorithm, an iterative time-domain channel estimation approach capable of using a priori information is proposed for orthogonal frequency division multiplexing (OFDM) systems in this letter: it outperforms its noniterative counterpart in terms of estimation accuracy as well as bit error rate (BER) performance. Numerical simulations demonstrate that an SNR gain of 1 dB at BER=10-4 with only one iteration and estimation mean square error (MSE) which nearly coincides with the Cramer-Rao bound (CRB) in the low SNR region can be obtained, thanks to the efficient use of a priori information.

In this paper, we design a new coded cooperation protocol utilizing superposition modulation together with iterative decoding/detection algorithms. The aim of the proposed system is to apply "dirty paper coding" theory in the context of half-duplex relay systems. In the proposed system, the node transmits a superposed signal which consists of its own coded information and other node's re-coded information. The destination node detects and decodes the signal using the received signals at two continuous time-slots with iterative decoding algorithm. Moreover, the destination node detects the received signal using the results of decoding, iteratively. This paper provides the outage probability of the proposed system under the assumption that the proposed system can ideally perform dirty paper coding, and it is shown from the comparison between outage probabilities and simulated results that the proposed system can get close to the dirty paper coding theory.

In this letter, we study the performance of the iterative receiver as applied to the space division multiplexing/orthogonal frequency division multiplexing (SDM/OFDM) systems. The iterative receiver under consideration employs the soft in/soft out (SISO) decoding process, which operates iteratively in conjunction with channel estimation for performing data detection and channel estimation at the same time. As opposed to the previous studies in which the perfect channel state information is assumed, the effects of channel estimation are taken into account for evaluating the performance of the iterative receiver and it is shown that the channel estimation applied in every iteration step of the iterative receiver plays a crucial role to warrant the performance, especially at a low signal-to-noise power ratio (SNR).

To achieve a sound field reproduction system, it is important to design multichannel inverse filters which cancel the effects of room transfer functions. The design method in the frequency domain based on the least-norm solution (LNS) requires less memory and less calculation than the design method in the time domain. However, the LNS method cannot guarantee the causality or stability of the filters. In this paper, a design method of a time-domain inverse filter using iterative processing in the frequency domain for multichannel sound field reproduction is proposed, and the result of numerical analysis is described. The proposed method can decrease the squared error of every control point by 3-12 dB. Furthermore, the sound reproduced by this method attains over 13 dB improvement in the segmental signal-noise ratio (SNR) compared with that designed by the LNS method for real environment impulse responses.