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.

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.