In this paper, we propose an improved MIMO successive interference cancellation (SIC) detector taking maximum likelihood (ML) criterion into account. Applying the ML criterion to multiple candidates obtained from all possible orderings of streams in the SIC, the proposed method can provide improved performance with complexity comparable to conventional ordered SIC for moderate number of spatial streams.

This paper proposes a dimension reduction maximum likelihood (ML) soft demodulator for multiple-input multiple-output (MIMO) system. The proposed dimension reduction soft demodulator reduces the dimension of the search space for minimum Euclidean distance calculation by exploiting ML hard detection algorithms and can achieve significantly lower complexity than existing optimal soft demodulators based on exhaustive or near-exhaustive search. The dimension reduction can be realized through the use of optimum MIMO hard detectors such as a sphere decoder, resulting in optimal performance. It can also be implemented using suboptimum low-complexity MIMO hard detectors such as various conventional MIMO equalizers, further reducing the complexity with slight performance degradation.

When a preamble signal is repeated multiple times in OFDM systems, we derive joint maximum likelihood (ML) estimation of the frame timing (FT) and carrier frequency offset (CFO). Unlike conventional estimators which use correlation of adjacent repetition patterns only or some specific sets of patterns, the joint ML estimation exploits correlation of any pair of repetition patterns, providing optimized performance. To reduce the implementation complexity involved in the joint ML estimation, we also propose a near-ML estimation method that separates estimation of the FT and the CFO. The performance of the proposed methods is verified by computer simulation.

Multiple-input multiple-output (MIMO) systems with hybrid automatic-repeat-request (HARQ) promises high throughput with high reliability. However, combining-scheme design for such systems faces challenges, the presence of interference and the existence of multiple signal-to-interference-and-noise power ratios (SINRs). To overcome these challenges, this paper suggests to design combining schemes with the objective of directly optimizing the log-likelihood ratio (LLR) values. Using this approach, this paper proposes several combining schemes and then analyzes them based on three key design factors: decoding performance, scalability, and memory requirement. Computer simulations under the IEEE 802.16e standard settings show the performances of the proposed combining schemes, which align well with the analysis. Based on the analyses and the simulation results, this paper suggests preferable combining schemes respectively for MIMO systems with HARQ-chase combining (HARQ-CC) and HARQ-incremental redundancy (HARQ-IR).

We propose and study a class of transmit beamforming techniques for systems with multiple transmit and multiple receive antennas with a per-antenna transmit power constraint. The per-antenna transmit power constraint is more realistic than the widely used total (across all transmit antennas) power constraint, since in practice each transmit antenna is driven by a separate power amplifier with a maximum power rating. Under the per-antenna power constraint, from an implementation perspective, it becomes desirable to vary only the phases (as opposed to both power and phase variation) of the signals departing from the transmit antennas. We name this class of techniques generalized co-phasing and formulate an optimization problem to calculate the transmit antenna phases. Furthermore, we propose five heuristic algorithms to solve the optimization problem. All the proposed algorithms except one are optimal for the case of two transmit antennas and an arbitrary number of receive antennas. For an arbitrary number of transmit and receive antennas, simulations indicate that the proposed algorithms perform very close to the optimal solution calculated through an exhaustive search of all possible transmit phases.

The implementation of decision feedback equalization-based receivers for MIMO systems that employ hybrid ARQ (HARQ) is considered. It is shown how the theory of MIMO decision feedback equalization can be applied to derive optimal DFE architectures for the case of MIMO HARQ. Incremental structures are proposed that reduce the implementation complexity of the DFE and the memory requirements at the receiver without loss in performance. Moreover, sub-optimal, post-equalization combining architectures are examined that simplify the receiver at the cost of performance loss.

Hybrid Automatic Repeat reQuest (HARQ) schemes are becoming common in wireless systems. In addition to the channel characteristics, several system design considerations need to be taken into account when choosing the particular HARQ scheme to be used. For SISO systems, it has been shown that the coding gain of Incremental Redundancy (IR-) HARQ over Chase Combining (CC-) HARQ may be offset by adverse fading channel conditions. In this paper it is shown that, for MIMO systems, IR- HARQ may also be affected by practical complexity constraints on the implementation of the receiver that dictate the use of bit-level combining. In some cases the coding gain advantage of IR-HARQ may be offset by the inability to perform symbol combining in the receiver as opposed to the case when CC-HARQ is employed. A system compliant with the IEEE 802.16e standard that employs a Convolutional Turbo Code is used to demonstrate an example of such performance loss.

For practical multi-input multi-output (MIMO) orthogonal frequency division multiplexing (OFDM) systems, it is important to develop transmit diversity schemes that are transparent to a receiver in that the receiver does not need to perform additional processing. In this paper, we examine one of the most widely-used receiver-transparent transmit diversity schemes - the cyclic delay diversity (CDD) scheme. It is shown that the CDD scheme effectively increases the delay spread of the channel and that the performance degrades under non-ideal channel estimation. As a remedy to the performance loss due to non-ideal channel estimation, a cluster-based transmit diversity scheme is proposed, which garners almost all the benefits of the CDD scheme without degrading the performance of a cluster- based channel estimation. The simulation results are provided in the realistic setting of IEEE 802.16e WiMAX systems.

This paper examines the detection problem of the preamble sequence index in the WiMAX system. The mobile station receiver knows all the possible preamble sequences and should estimate which preamble sequence has been transmitted from the base station. Since the preamble in the orthogonal frequency division multiplexing (OFDM) transmission is usually the first received symbol, the channel is unknown to the receiver, which makes the problem of preamble sequence estimation complicated. In this paper, this problem is addressed by developing the joint maximum likelihood (ML) estimator of the preamble sequence and the channel. A simple decoupled estimator and a minimum mean square error (MMSE) estimator are also presented as benchmarks for the joint ML estimator. Then it is shown how the joint ML estimator can be used for the segment detection. Since the joint ML estimator can be computationally complex in its general form, low-complexity algorithms are developed depending on the type of pilot subcarrier locations for general OFDM systems including WiMAX. The simulation results show that the joint ML estimator detects the preamble sequence index very well in the absence of the channel knowledge.

This paper examines the detection problem of the preamble sequence index in the orthogonal frequency division multiplexing (OFDM) system. The receiver knows all the possible preamble sequences and should estimate which preamble sequence has been transmitted. Preamble sequence detection is straightforward with the knowledge of channel conditions. However, since the preamble OFDM symbol is usually the first symbol that is received, the channel is unknown to the receiver, which makes this problem complicated. In this paper, the joint maximum likelihood (ML) estimator of the preamble sequence and the channel is derived and compared with a simple decoupled estimator. The estimators are applied to IEEE 802.16e systems, and it is shown by simulation that the joint ML estimator detects the preamble sequence index very well in the absence of the channel knowledge.

This paper proposes and analyzes receiver schemes for multiple-input multiple-output (MIMO) systems with hybrid automatic-repeat-request (HARQ). It is shown by means of analysis as well as computer simulations that the proposed receiver schemes have optimal decoding performance in the sense that all the relevant information is fully used. Also complexity-wise, the proposed receiver schemes are found beneficial when designing MIMO systems, because the same decoding architecture can be re-used regardless of the number of retransmissions.

For orthogonal frequency division multiplexing (OFDM) systems, the carrier frequency offset (CFO) between the carrier frequencies of the received signal and the receiver local oscillator is represented as the sum of the integer part and the fractional part by normalizing the CFO with subcarrier spacing. This paper focuses on the pilot-aided estimation of the channel gain and the integer part of the normalized CFO OFDM systems. For multipath channels, a simple estimator of the channel gain and the integer CFO is described, and the joint maximum likelihood (ML) estimator is developed. It is proved that the simple estimator becomes equivalent to the joint ML estimator when two OFDM symbols are observed. For the AWGN channel, this paper derives both the ML estimator of the integer CFO only and the joint ML estimator of the integer CFO and the phase offset. It is proved that the integer CFO estimation part of the joint ML estimator is the same as the ML estimator of the integer CFO only, and it is also shown by simulation that the ML estimator outperforms existing estimators significantly.

This letter analyzes the effect of the carrier frequency offset on orthogonal frequency division multiplexing (OFDM) systems for multipath fading channels. A simple approximate expression for the average signal-to-noise ratio (SNR) is derived. This approximate expression is shown to be an upper bound of the average SNR for flat fading channels and an exact expression for the AWGN channel. The approximate average SNR expression is validated using Monte Carlo simulation for both flat fading channels and frequency-selective fading channels

This paper analyzes the effect of phase noise on orthogonal frequency division multiplexing (OFDM) systems with a focus on time-direction (TD) differential receivers. Both Wiener phase noise and wide-sense stationary (WSS) phase noise are considered. A signal-to-noise ratio (SNR) expression for TD differential receivers is derived in the presence of phase noise and is compared to that for coherent receivers, common phase error correction (CPEC) receivers, and frequency-direction (FD) differential receivers. It is shown that Wiener phase noise affects TD differential receivers more than CPEC receivers and FD differential receivers. On the other hand, for WSS phase noise, the performance of TD differential receivers depends on the shape of the power spectral density of the phase noise and the subcarrier bandwidth. When the power of the phase noise in one subcarrier bandwidth is much smaller than the total power of the phase noise, the performance of TD differential receivers is comparable to that of FD differential receivers

In orthogonal frequency division multiplexing (OFDM) systems, the carrier frequency offset can he normalized by the subcarrier bandwidth and then he divided into the integer and the fractional part. This paper focuses on the estimation of the integer carrier frequency offset and derives approximate maximum likelihood (ML) estimators in the presence and in the absence of pilot subcarriers. Simplified approximate ML estimators with reduced complexity are also proposed. The performance of the proposed estimators is compared with that of existing estimators, and it is shown by simulation that the proposed estimators outperform existing estimators.

This paper presents blind symbol timing estimators in orthogonal frequency division multiplexing (OFDM) systems for the additive white Gaussian noise (AWGN) channel. Depending on the performance measure, different optimal estimators can be devised. The existing maximum likelihood estimator is optimal in the sense that it maximizes the probability that the symbol timing estimation error is zero, but it does not guarantee that the error will be small when the estimate is incorrect. When maintaining small error magnitude is critical, the minimum mean square error estimator can be developed with the objective of minimizing the mean square error. In some practical scenarios, the estimation error needs to be defined as the distance between the starting point of the fast-Fourier transform window and the region consisting of the cyclic prefix samples and the first useful sample. With this modified definition of the estimation error, the minimum failure probability estimator and the minimum modified mean square error estimator can be derived. The performance of the proposed estimators is compared with that of the maximum likelihood estimator using simulation.

The paper presents an exact analysis of the effect of the carrier frequency offset on orthogonal frequency division multiplexing (OFDM) systems for a general multipath fading channel. As is well known, the carrier frequency offset attenuates the desired signal and causes intercarrier interference, thus reducing the signal-to-noise ratio (SNR). The SNR degradation due to the carrier frequency offset is evaluated by deriving the exact SNR expression in the presence of the carrier frequency offset. The SNR analysis can be used for the design of a practical OFDM system in determining how small the frequency offset should be in order to maintain the SNR degradation to negligible levels.

In this paper we investigate the application of Unequal Error Protection (UEP) channel coding schemes to the Transform Predictive Coding (TPC) audio coding paradigm. The TPC paradigm has natural perceptual ordering of the bitstream that provides a good match with the UEP concept. Also introduced is the general idea of Mode Adaptive UEP schemes that adapt to the differing bit-error sensitivity profiles of signal adaptive multimode source-coders. By doing this the decoded signal quality is optimized jointly over the input signal, the channel conditions and the bit-error sensitivity profile of the source coder.

The move to digital is a natural progression taking place in all aspects of broadcast media applications from document processing in newspapers to video processing in television distribution. This is no less true for audio broadcasting which has taken a unique development path in the United States. This path has been heavily influenced by a combination of regulatory and migratory requirements specific to the U.S. market. In addition, competition between proposed terrestrial and satellite systems combined with increasing consumer expectations have set ambitious, and often changing, requirements for the systems. The result has been a unique set of evolving requirements on source coding, channel coding, and modulation technologies to make these systems a reality. This paper outlines the technical development of the terrestrial wireless and satellite audio broadcasting systems in the U.S., providing details on specific source and channel coding designs and adding perspective on why specific designs were selected in the final systems. These systems are also compared to other systems such as Eureka-147, DRM, and Worldspace, developed under different requirements.

Hybrid in band on channel (IBOC) digital audio broadcasting (DAB) simultaneously with analog amplitude modulation (AM) has been proposed as a hybrid solution to digital audio broadcasting in the AM band. Adding digital transmission in the crowded AM band is a challenging proposition. To achieve FM like audio quality, an audio coder rate of 32-64 kbit/s may be required. One of the currently proposed hybrid IBOC-AM systems is 30 kHz wide. Severe second adjacent interference may occur in certain geographical areas. For coping with such harsh transmission conditions, we present a solution based on embedded/multi-descriptive audio coding with matched multistream transmission in separate frequency bands. With loss of one frequency band, the embedded system blends to a lower audio coder rate with still a better quality than analog AM. The non-embedded system without multi-stream transmission fails catastrophically causing a severe discontinuity in quality while blending directly to analog AM.

High quality multimedia streaming applications over the Internet require very low packet loss rates. The Internet is characterized by long bursts of packet losses and delays. A large receive buffer can be used to mitigate the effects of packet losses and delays. However, a large receive buffer introduces a large delay in the playback of a packet. This large delay could be annoying at the start of a program or during switch over to another channel in a multi-channel broadcast. We introduce a separate low-delay tuning stream to address this start-up problem. In the steady state, this tuning stream is synchronized appropriately with the high-delay steady state stream to give an enhanced composite signal.

In state-of-the-art digital communication systems, channel estimation and/or equalization is an indispensable part of the system. Due to limited resources, only relatively simple algorithms with low complexity can be applied for channel estimation and equalization in high data-rate systems. In fact, the complexity of these algorithms can become prohibitive for very high bit-rate applications, for example, several 100 Msymbols-per-second. We explore the signal constellation structure of practical digital communication systems and describe an efficient computation technique that not only eliminates multiplications but also minimizes the number of addition operations required to implement channel estimation algorithms such as the least mean squares (LMS) estimation and equalization algorithms such as the maximum likelihood (ML) sequence detection using the Viterbi algorithm. This new technique preserves the numerical precision of the algorithms while it reduces their complexity dramatically.

In the Viterbi algorithm for decoding convolutional and trellis codes, squared Euclidean distance is the optimum branch metric for decoding sequences transmitted in additive white Gaussian noise channels and in Rayleigh fading channels when the channel state information is appropriately included. For high bit-rate applications; multiple parallel branch metric computation units may be required. Duplication of the multipliers or large look-up tables to obtain the squared distances can increase the decoder complexity significantly. Furthermore, the multipliers and the delay occurred in accessing large look-up tables can become the bottleneck for a fully pipelined Viterbi decoder. In this paper, we show that linear distances can be used to represent the branch metrics, without compromising the Viterbi decoder performance. Thus, adders rather than multipliers can be used to compute the branch metrics and this breaks the potential bottleneck in the Viterbi decoder.

Several systems are being considered for digital audio broadcasting (DAB) and some of these systems will be deployed for commercial use. In this paper, we consider a code division multiplex (CDM) scheme for satellite DAB. For diversity purposes, this system makes use of at least two satellites, and in urban areas, where the satellite signals are shadowed by large structures, a terrestrial network is employed. We present a forward error protection (FEC) scheme that is robust under a variety of channel conditions, especially in cases where the received signal is severely attenuated because of highway underpasses and tunnels. This FEC scheme makes use of time diversity, and this introduces a large delay. Such delays will not affect the performance of broadcast systems; however, the delay will be an issue during tuning when a switch to another channel occurs. We introduce a separate low-delay low-rate tuning channel to facilitate tuning. Extensive simulation results are given to examine the performance of the system, and it is shown that about 70 audio channels, each with a rate of 96 kbit/s, can be satisfactorily supported in a bandwidth of approximately 12.5 MHz in most of the channel conditions considered.