We study the effects of channel estimation errors on bit error rate (BER) performance of a DFT-precoded OFDM system with frequency domain linear minimum mean square error (MMSE) equalizer, by analytically studying the effects of channel estimation errors on the mean square error (MSE) of the equalizer output, which can be related to the BER performance using a Gaussian approximation. We first obtain an approximate expression for the BER performance in terms of the MSE of the equalizer output with channel estimation errors, which is found to be a sum of the equalizer output MSE for the known channel frequency response and the scaled MSE of the channel frequency estimates. For a fixed MSE of channel frequency estimates, the degradation of the equalizer output signal-to-noise ratio (SNR) increases as Es/Noincreases. We also propose a semi-analytical approach to obtain the BER performance, by modeling the frequency channel estimates as the actual channel estimates plus a Gaussian random variable with zero mean and a variance of the averaged MSE of the channel frequency estimates. Simulation results show that both proposed approaches can accurately estimate the BER performance of DFT-precoded OFDM systems with channel estimation errors.

The large instantaneous power fluctuations of orthogonal frequency division multiple access (OFDMA) signals necessitate significant power backoff to minimize the effects of nonlinear power amplifier distortion. Is there an optimum linear block precoding that significantly reduces the instantaneous power variance of OFDMA signals? For contiguous and block-interleaved non-contiguous frequency mappings we find that there is a power variance-minimizing linear block precoding which requires a slightly lower power backoff than that for DFT precoding (which in the form of SC-FDMA, is used for the uplink signaling in the 3GPP-LTE standard). The optimum precoding is not unique, and in its simplest form it imposes only mild linear filtering and a moderate noise enhancement penalty if the receiver uses linear equalization.

Is there an optimum linear block precoding for orthogonal frequency division multiple access (OFDMA) signals that significantly reduces their instantaneous power variance? We explore this question for linear block-precoded OFDMA signals with contiguous frequency mapping. We find that there is an optimum linear block precoding, but its advantage over DFT precoding (such as that used for the uplink signaling in the 3GPP-LTE standard) is rather small.

Phase noise can severely impair transmission, especially in uplink space division multiple access (SDMA) scenarios. We derive a soft-decision-directed phase noise estimation and compensation algorithm in combination with iterative block decision feedback equalization (IBDFE) for uplink SDMA single carrier systems, such as those proposed in the 3GPP-LTE standard. The algorithm generalizes earlier algorithms for non-SDMA (single-user) systems. It is shown by simulation to be effective and relatively simple to implement.

In recent years single carrier modulation (SCM) has again become an interesting and complementary alternative to multicarrier modulations such as orthogonal frequency division multiplexing (OFDM). This has been largely due to the use of nonlinear equalizer structures implemented in part in the frequency domain by means of fast Fourier transforms, bringing the complexity close to that of OFDM. Here a nonlinear equalizer is formed with a linear filter to remove part of intersymbol interference, followed by a canceler of remaining interference by using previous detected data. Moreover, the capacity of SCM is similar to that of OFDM in highly dispersive channels only if a nonlinear equalizer is adopted at the receiver. Indeed, the study of efficient nonlinear frequency domain equalization techniques has further pushed the adoption of SCM in various standards. This tutorial paper aims at providing an overview of nonlinear equalization methods as a key ingredient in receivers of SCM for wideband transmission. We review both hybrid (with filters implemented both in time and frequency domain) and all-frequency-domain iterative structures. Application of nonlinear frequency domain equalizers to a multiple input multiple output scenario is also investigated, with a comparison of two architectures for interference reduction. We also present methods for channel estimation and alternatives for pilot insertion. The impact on SCM transmission of impairments such as phase noise, frequency offset and saturation due to high power amplifiers is also assessed. The comparison among the considered frequency domain equalization techniques is based both on complexity and performance, in terms of bit error rate or throughput.

Relaying and orthogonal frequency division multiple access (OFDMA) are the accepted technologies for emerging wireless communications standards. The activities in many wireless standardization bodies and forums, for example IEEE 802.16 j/m and LTE-Advanced, attest to this fact. The availability or lack thereof of efficient radio resource management (RRM) could make or mar the opportunities in these networks. Although distributed schemes are more attractive, it is essential to seek outstanding performance benchmarks to which various decentralized schemes can be compared. Therefore, this paper provides a comprehensive centralized RRM algorithm for downlink OFDMA cellular fixed relay networks in a way to ensure user fairness with minimal impact on network throughput. In contrast, it has been observed that pure opportunistic schemes and fairness-aware schemes relying solely on achievable and allocated capacities may not attain the desired fairness, e.g., proportional fair scheduling. The proposed scheme is queue-aware and performs three functions jointly; dynamic routing, fair scheduling, and load balancing among cell nodes. We show that the proposed centralized scheme is different from the traditional centralized schemes in terms of the substantial savings in complexity and feedback overhead.

The provision of very high data rates in a ubiquitous and fair manner throughout the service area is a great challenge for 4G and beyond-4G wireless networks. Towards that end, the deployment of fixed relays by the operators has become an accepted radio access network concept in various standardization activities including LTE-A and 802.16m. In this paper, we present a fair resource allocation scheme that can significantly reduce the co-channel interference and improve spectrum utilization in OFDMA-based multi-cellular networks enhanced by fixed relays. The major goal is to obtain an effective throughput-fairness trade-off through the combination of dynamic resource allocation, in-cell routing, traffic diversity, and history of users' past service. The proposed scheme achieves substantial fairness, throughput, and latency improvements in comparison to the reference schemes as well as to another scheme proposed, earlier by the authors. Numerical results indicate that the proposed scheme provides an almost ubiquitous high-data-rate service to users, irrespective of their locations (including cell-edge users), with a minimal sacrifice in the overall network throughput.

Relaying and orthogonal frequency division multiple access (OFDMA) are the accepted technologies for emerging wireless communications standards. The activities in many wireless standardization bodies and forums, for example IEEE 802.16 j/m and LTE-Advanced, attest to this fact. The availability or lack thereof of efficient radio resource management (RRM) could make or mar the opportunities in these networks. This paper therefore provides a comprehensive RRM algorithm for OFDMA-based multi-cellular fixed relay networks in a way to ensure fairness among users with minimal impact on the network throughput (in contrast, pure opportunistic RRM techniques always favor users with good channel conditions). Unlike the majority of works in the literature, our proposed scheme is queue-aware and jointly performs routing, fair scheduling, and load balancing among cell nodes. The routing strategy has inherent learning ability and it dynamically converges to better routes.

This paper derives bounds on the diversity- multiplexing tradeoff of wireless relay networks with arbitrary link connectivity between cooperating terminals. The derived bounds are applicable when there are single and multiple antennas per terminal. Two classes of relaying method are analyzed, those requiring all cooperating terminals to correctly decode the transmitted signal in order for the destination to correctly decode (comprehensive decoding), and those requiring only a subset of cooperating terminals to correctly decode the transmitted signal in order for the destination to correctly decode (destination decoding). It is shown that the diversity- multiplexing tradeoff of comprehensive decoding is constrained by the minimum number and arrangement of incident inter- terminal antenna links across all receiving terminals in the network, and that the diversity-multiplexing tradeoff of destination decoding is constrained by the minimum number and arrangement of incident inter-terminal antenna links across all cut sets in the network. Results for the complexity of the involved minimization algorithms are provided.

SC modulations (single-carrier) with FDE (frequency-domain equalization) are promising candidates for future broadband wireless systems. These modulations allow excellent performances in severely time-dispersive channels, provided that accurate channel estimates are available at the receiver. For this purpose, pilot symbols and/or training sequences are usually multiplexed with data symbols. Since this leads to overhead and some spectral degradation, the use of superimposed pilots (i.e., pilots added to data) was recently proposed. In this paper we consider SC-FDE systems where the channel estimation is based on superimposed pilots. Since the interference levels between data and pilots might be very high, we propose an iterative receiver with joint equalization and channel estimation. Our performance results show that the use of superimposed pilots, combined with the proposed receiver, allows performances close to the case with perfect channel estimation, even for severely time-dispersive channels.

In this paper we analytically evaluate the performance of turbo frequency domain equalizers when the frequency selective channel is estimated erroneously. We generalize the bit error rate transfer chart method proposed with the assumption of perfect channel information to the case with imperfect channel state information to analyze the performance of a turbo frequency domain equalizer. Through a comparative performance evaluation we show that the performance achieved by simulations is consistent with the one predicted by the bit error rate transfer chart. By this analysis tool we investigate the sensitivity of both adaptive and non- adaptive schemes to the processing gain of channel estimation. We show that the former case is less sensitive to channel estimation error.

We consider two techniques of in-band frequency domain multiplexed (FDM) pilots using interleaved frequency domain multiple access (IFDMA) signal with a Chu sequence for DFT-precoded OFDM (or single-carrier (SC)) systems. One, called frequency domain superimposed pilot technique (FDSPT), superimposes pilot tones onto scaled or deleted data tones, which preserves spectral efficiency at the expense of a slight performance loss. The other, called frequency expanding technique (FET), multiplexes pilot tones by displacing data tones, which slightly reduces spectral efficiency. Using FDM pilots in SC systems facilitates flexible and efficient assignment of signals to available spectrum. We propose an iterative frequency domain decision-directed interference cancellation technique to reduce the intersymbol interference level of SC signals with FDSPT pilots (resulting from the suppression of data tones). Moreover, we propose a low complexity frequency domain iterative decision-directed channel estimation (IDDCE) technique for SC systems using FDM pilots. Using IDDCE, the frame error rate (FER) performance for coded SC systems using FET and FDSPT pilots with interference cancellation is found to be about 0.2 dB and about 0.5 dB, respectively, away from the FER performance with known channel frequency response at FER=10-2. FDSPT pilots can also be used for OFDM systems with channel coding. It is found that an extra 1 dB of SNR is required at FER=10-2,compared with that using the conventional FET pilots for OFDM systems.

In this paper we propose a synchronization method to compensate for the effect of frequency offset and phase noise generated by local oscillator instabilities for a Turbo Frequency Domain Equalizer (TFDE). TFDE is considered as an efficient equalization method which benefits from both good performance of iterative systems and reasonable complexity due to performing equalization in the frequency domain. We propose a joint turbo equalization and synchronization method in which the feedback information, generated iteratively by the decoder, is used for synchronization. As iterations continue, if the feedback information becomes more reliable, the performance of the synchronization method is continually improved. We propose two methods based on different criteria when the system suffers from only frequency offset. We also suggest a method based on Maximum A Posteriori (MAP) criterion in the presence of both frequency offset and phase noise which can be considered as a decision directed phase lock loop. These methods achieve considerable performance improvement with reasonable complexity.

By the generalized multi-carrier (GMC) principle a unified framework to describe various multi-carrier as well as single carrier approaches is established. In this paper the GMC principle is extended to pilot design and channel estimation, so that a unified description of pilot aided channel estimation (PACE) by interpolation in time and frequency is established. This applies to frequency domain pilots which are embedded in the GMC signal, as well as pilots sequences time multiplexed with data-bearing GMC blocks. A comparative performance evaluation of the two different GMC variants OFDM and single carrier with various pilot allocation schemes is carried out, also taking into account practical constraints such as the peak to average power ratio (PAPR) and required power amplifier back-off. It is found that a serial modem with time or frequency multiplexed pilots could be designed with a power amplifier with about 2 dB lower maximum power rating than that of a corresponding OFDM modem.

A low complexity frequency domain iterative decision-directed channel estimation (FD-IDDCE) technique for single-carrier (SC) systems with frequency domain multiplexed (FDM) pilots is proposed. The tentative hard decisions from either the frequency domain equalizer output or the Viterbi decoder output are used as extra pilots to further improve the initial frequency channel estimates using a cascaded 2 times 1D Wiener filter. The issue of noise enhancement when finding the least square (LS) estimates of the channel frequency response (CFR) using the tentative decisions in the frequency domain is overcome by the so called frequency replacement algorithm, which replaces the noise enhanced LS estimates of the CFR with the corresponding channel frequency estimates in the previous iteration, by comparing with a threshold. Using the proposed FD-IDDCE, the frame error rate (FER) performance for coded SC systems using FDM pilots with frequency expanding technique and frequency domain superimposed pilot technique was found to be about 0.4 dB and about 1 dB away from the FER performance with known CFR at FER=10-2.

This paper analyzes the performance of turbo frequency domain equalization (TFDE) with 16-QAM by bit error rate transfer (BERT) chart. We derive the BERT chart for the two cases in which the equalizer adapts or does not adapt from iteration to iteration. In the first case, the equalizer parameters change during iterations based on the feedback information and in the latter case they do not change. In the BERT chart analysis of TFDE, we theoretically find the mean and variance of the Gaussian-like distribution of the equalizer output. The mean and variance of the equalizer output depend on the channel parameters, channel signal to noise ratio and the decoder bit error rate in the previous iteration. Finally we compare the BERT chart results with simulation results and show that the predicted results are fairly accurate.

In this paper, the issue of channel estimation in Single Carrier Frequency Domain Equalization (SC-FDE) uplink Space Division Multiple Access (SDMA) systems in the presence of intra and inter-cell interference is addressed. It is shown that in such an interference-prone system, a combination of orthogonal pilots, spatial processing and Iterative Channel Estimation (ICE) is effective. Adaptive Iterative Block Soft Decision based Feedback Equalizer (IBDFE) with Parallel Interference Cancellation (PIC) assisted Decision Feedback Iterative Channel Estimation is employed. It is shown that such a combination is a promising way to suppress uplink inter-user interference.

This paper derives bounds on the diversity order and high SNR probability of outage behavior of wireless relay networks with arbitrary link connectivity between cooperating terminals. Single and multiple antennas per terminal are considered. Two general codebook generation schemes are analyzed and compared. Common codebook generation involves the source and all relays sharing a common randomly generated codebook, and encompasses many practical encoding schemes, including both repetition coding and space-time coding. Independent codebook generation involves the source and all relays employing independent randomly generated codebooks, and provides an information theoretic lower bound on the probability of outage of all achievable codebook generation schemes. The results indicate that although independent codebook generation does not offer any diversity gain over common codebook generation, it can offer a significant probability of outage improvement under certain conditions.

We investigate the feasibility of applying the peak-to-average power ratio (PAPR) reduction method using pilot sequences, originally proposed for orthogonal frequency division multiplexing (OFDM) signals, for single-carrier (SC) signals with frequency domain multiplexed (FDM) pilots. The idea is to select the FDM pilot sequence with which the transmitted signal produces the lowest PAPR. We also investigate the applicability of the sum of square error (SSE) selection rule for high order modulation of SC signals. The SSE rule selects the pilot sequence which produces the minimum SSE between the transmitted signal and a pre-defined threshold, proportional to the saturation level of a high power amplifier (HPA). It is found that for both SC and OFDM systems, the PAPR reduction capabilities of orthogonal Walsh-Hadamard (W-H) sequences and cyclic shifted Chu (CS-Chu) sequences are similar for small block size, but not for large block size. Using CS-Chu sequences produces better PAPR reduction capability. With an appropriate choice of the value of input backoff power of a HPA, the SSE selection rule produces similar results as that of the minimum PAPR selection rule. The effects of out-of-band radiation for SC and OFDM signals with PAPR reduction using FDM pilot sequences after HPA depends on the amount of non-linearity portion of the HPA. The out-of-band radiation improvement is obvious for a HPA that approximates a linear clipper.

A maximum-likelihood channel estimator for the orthogonal frequency division multiplexing communication environments, in the presence of interference is discussed here. In a training-based scenario, the channel is estimated based on pilots that precede the transmission of the information. To reduce the number of estimation parameters, the channel is estimated iteratively in time-domain. Since interference from other users provide no useful information, parameters of the interference are neither estimated nor the effect of the interference neglected, instead interference along with Gaussian noise is perceived as non-Gaussian noise process. The algorithm assumes no a priori knowledge about the interfering channel and signal at the receiver, further no assumption on the statistical properties of the interferer is assumed, which makes this algorithm robust. The estimated channel information along with the estimated distribution are then utilised to equalise the subsequent data blocks. The strength of the algorithm is in its robustness to both synchronous and asynchronous interference, which is confirmed by the simulation results for both flat and multipath fading channels in presence of synchronous and asynchronous interference.

This paper considers the ways that cooperating terminals can be connected to each other in wireless relay networks and the constraints imposed by the availability of different system resources. A framework is developed that exposes the relationship between constraints on available system resources and the achievable combinations of communication links between cooperating terminals. Cooperative connectivity models defined by the achievable combinations of links are derived, associated with their minimum cost constraint sets, and mapped to diversity techniques presented in the literature. The constraints considered are the available number of orthogonal relaying channels, the ability of terminals to diversity combine signals on a single common channel, the ability of terminals to diversity combine signals on orthogonal channels, the ability of terminals to transmit signals on multiple orthogonal channels, and the ability of terminals to cancel the effects of interhop interference.

In this paper, a new class of equalization and channel estimation techniques, using the turbo frequency domain equalization (TFDE), is presented as a promising low-complexity detection method for single-carrier broadband wireless transmissions. Serial modulation (SM), being a direct counterpart of the well-known OFDM modulation, is receiving considerable attention recently, owing to the fact that it delivers comparable performance as OFDM while avoiding the problem of high peak-to-average power ratio. When frequency domain equalization (FDE) is applied, the complexity requirement is low and it becomes feasible to employ iterative processing which relies on decision feedback. This paper considers the turbo principle applied jointly to FDE, channel decoding and channel estimation. The result of this work is a set of effective iterative algorithms which may bring about 2-3 dB improvement over the linear FDE method. Furthermore, it is shown that they can provide performance comparable to the time-domain turbo equalization methods but with lower complexity. We then reach the conclusion that SM-based transmission with TFDE is a suitable technology for next generation wireless systems

This paper1 presents a Turbo Equalization system using soft interference cancellation in the frequency domain, in which the filter is updated during the iterations with respect to the soft information it receives from the decoder. Simulation results confirm that this method outperforms the suggested one in [4,ll] while it does not add much complexity to the system. We also investigate the effect of two coding schemes, LDPC code and convolutional code, on the performance of Turbo Frequency Domain Equalization (TFDE). TFDE with these codes are compared in terms of performance and computational complexity. Finally we suggest using an LDPC based frequency domain turbo equalizer due to its good performance/complexity tradeoff.

We investigate channel estimation for single-carrier-frequency domain equalization (SC-FDE) system using the techniques typically used for an orthogonal frequency domain multiplexing (OFDM) system. Two techniques of frequency domain multiplexed (FDM) pilot insertion using interleaved frequency domain multiple access (IFDMA) signal with a Chu sequence are considered. One called frequency domain superimposed pilot technique (FDSPT) scales data-carrying tones and then superimposes them with pilot tones. This technique preserves spectral efficiency at the expense of performance loss. The other, called frequency expanding technique (FET), shifts groups of data frequencies for multiplexing of pilot tones at the expense of spectral efficiency. Our results show that both techniques increase peak to average power ratio (PAPR) although it is still lower than that of an OFDM system. The application of FDSPT is limited by the pilot overhead ratio, resulting from the removal of data frequencies for pilot frequencies. It is shown that channel estimation using conventional time domain multiplexed pilots and FET pilot tones produce the same BER, while the FDSPT requires about 1.5 dB more power for the same performance. Using FDM pilots in SC system facilitates flexible and efficient assignment of signals to available spectrum.

This paper compares peak to average power ratio (PAPR) of multi carrier code division multiple access (MC-CDMA) and serial modulation-CDMA (SM-CDMA) with each other and with corresponding non-spread spectrum techniques which are orthogonal frequency division multiplexing (OFDM) and serial modulation (SM). In this comparison we consider the effect of two spreading types- binary and complex spreading, and equal-level and multi-level constellation on the amplitude distribution and out-of-band radiation. While MC-CDMA and OFDM have similar amplitude distribution, we show that the PAPR of SM-CDMA can be larger than that of SM. If the spreading is binary and the constellation is multi-level, this effect is so powerful that the PAPR of SM-CDMA becomes similar to that of MC-CDMA. Finally we use a modified selected mapping (SLM) algorithm to decrease the PAPR of SM-CDMA so that it would be equal to that of SM. Therefore SM-CDMA has lower PAPR if we use it together with modified SLM, and the system will need amplifiers with lower power back-off values

In this paper we analyze the performance of a turbo frequency domain equalizer using the BER transfer chart. This tool evaluates the signal to noise ratio improvement at the decoder input during iterations. We derive a formula for the variance of the equalizer output at each iteration as a function of the error probability of the previous iteration. By defining an equivalent SNR based on the equalizer output mean and variance, and knowing the decoder bit error rate curve, we are able to evaluate the bit error rate of the decoder output in each iteration. Compared to the initially proposed BER transfer charts, this method gives us a more accurate curve for the equalizer and follows the dynamic of the process. Simulation results show that this method can predict the performance of the system with reasonable accuracy.

In this paper, we consider the uplink of SC-based (single-carrier) block transmission systems where the transmission bandwidth associated to each channel, exceeds the symbol rate and we can have significant overlapping between adjacent channels. We present an iterative frequency-domain receiver for joint equalization and ACI (adjacent channel interference) suppression taking advantage of spectral correlations for the separation of cyclostationary signals. Our performance results show that the BER (bit error rate) performances with strong ACI levels can be similar to the ones without ACI. Moreover, after a few iterations these performances can be close to MFB (matched filter bound)

The paper considers the impact of various system resource constraints on the connectivity and performance of relaying channels. This consideration is motivated by recent findings indicating that the performance of wireless relaying networks can be increased through the application of distributed spatial diversity techniques (e.g., multi-user diversity, multihop diversity, cooperative diversity) that rely on the mesh connectivity between wireless terminals. Relevant system resource constraints are introduced, constraint combinations are analyzed, and resultant system connectivity models are derived and compared via simulation. The constraints considered are the available number of orthogonal relaying channels, the ability of relays to diversity-combine signals, the ability of the destination to diversity-combine signals, the ability of receivers to diversity-combine signals on multiple orthogonal channels, and the ability of transmitters to transmit signals on multiple orthogonal channels. The analysis of these system resource constraints is applicable to both amplified (non-regenerative, amplify-and-forward) relaying and decoded (regenerative, decode-and-forward) relaying.

This paper considers the aggregate signal to noise ratio of amplified relaying channels, where the term aggregate refers to the inclusion of propagated noise terms generated as relaying terminals amplify both the information and noise portions of received signals. This consideration is motivated by recent findings for wireless relaying networks indicating that the performance of amplified relaying can approach and in some cases exceed the performance of decoded relaying. Expressions for the aggregate signal to noise ratio are developed for general amplified relaying channels with a given set of source, destination, and relaying terminals, link connectivity link attenuation, transmit power, and receiver noise. These expressions provide a method for analyzing the impact of varying the link connectivity or power allocation for a given set of terminals, and support the comparison of amplified relaying channels with decoded relaying channels.

We present a multiple access scheme for the uplink of broadband wireless systems. We consider SC (single carrier) based block transmission, with all users transmitting continuously, regardless of their data-rate. The transmitted signals can have very low envelope fluctuations. Moreover, the different users remain orthogonal, even for severe time-dispersive channels. By employing IB-DFE (iterative block decision feedback equalization) techniques we can have detection performances close to the matched filter bound, and even for fully loaded systems and severe time-dispersive channels.

The concept of relaying is a promising solution for the challenging throughput and high data rate coverage requirements of future wireless cellular networks. In this paper, we demonstrate that the area that a single BS (base station) can provide high data rate coverage can significantly be increased by the employment of digital fixed relays without any penalty in capacity. This network architecture, which allows two-hop links, is expected to facilitate cost-efficient high data rate coverage in beyond-3G cellular wireless networks. In particular, we considered the downlink of a non-CDMA network where 6 digital fixed relays are placed around each BS in a hexagonal layout. The improved two-hop links can be exploited to yield higher throughput through the use of adaptive modulation and coding.

A novel iterative frequency domain equalization method is proposed for single-carrier (SC) based wireless systems operating over a time- and frequency-selective (or doubly-selective) channel. We adopt the basis expansion channel model (BEM) to approximate the doubly-selective channel with a set of Fourier exponentials. Using BEM, time-varying multipath interference is translated into inter-carrier-like interference in the frequency domain. We design a low-complexity iterative method which relies on a sliding-window-based MMSE widely-linear filter with inter-carrier interference cancellation, and iterated time-domain soft-decisions. Simulations are shown to justify the validity of our approach and the results indicate that this approach outperforms the high-complexity linear block MMSE equalizer and other iterative approaches of similar complexity.

The use of multiple antennas at the BS (base station) allows significant improvements on the spectral efficiency of wireless communication systems by increasing the number of simultaneous users in a given cell. We introduce a new iterative multiuser detection scheme for systems requiring high-rate transmission in severe time-dispersive channels. We consider the use of single-carrier modulation combined with frequency-domain equalization techniques, which are known to be excellent candidates for severe time-dispersive channels. The BS employs multiple antennas and consists of an iterative LST (layered space-time) receiver combined with frequency-domain equalization techniques. Our performance results show that the proposed receiver structure has excellent performance, which can be very close to the matched filter bound, even for severe time-dispersive channels and in the presence of strongly interfering channels.

This work presents a theoretical analysis of the matched-filter bound on N-branch space-time processing receivers in a frequency-selective correlated fading environment. The development is based on a Karhunen-Loeve expansion in the frequency domain. Although many other works are based on similar expansions, we present a more general analytical framework in the frequency domain covering Rayleigh and Rician fading scenarios, with and without branch correlations and / or unequal branch powers. The Rician fading scenario is approximated using a Nakagami-m distribution. The branch correlations and unequal branch powers are dealt with through a simple and novel device, the concatenated array-equivalent channel. Likewise, a novel approach is presented which reduces the Rician fading case to an equivalent Rayleigh-fading system. Furthermore, results are presented for different array topologies where correlation characteristics are obtained with the help of a measurement-validated correlation model situated in the LMDS (local multipoint distribution service) band, i.e. the carrier frequency used in the measurements was 29.5 GHz.

This paper presents theoretical characterizations and analysis for the physical layer of multihop wireless communications channels. Four channel models are considered and developed: the decoded relaying multihop channel; the amplified relaying multihop channel; the decoded relaying multihop diversity channel; and the amplified relaying multihop diversity channel. Two classifications are discussed: decoded relaying versus amplified relaying, and multihop channels versus multihop diversity channels. The channel models are compared, through analysis and simulations, with the "singlehop" (direct transmission) reference channel on the basis of signal-to-noise ratio, probability of outage, probability of error, and optimal power allocation. Each of the four channel models is shown to outperform the singlehop reference channel under the condition that the set of intermediate relaying terminals is selected intelligently. Multihop diversity channels are shown to outperform multihop channels. Amplified relaying is shown to outperform decoded relaying despite noise propagation. This is attributed to the fact that amplified relaying does not suffer from the error propagation which limits the performance of decoded relaying channels to that of their weakest link.

This paper presents a theoretical analysis of the matched-filter bound on N-branch space-time processing receivers in a frequency-selective correlated fading environment. The development is based on a Karhunen-Loe/spl grave/ve expansion in the frequency domain. Although many other works are based on similar expansions, we present a more general analytical framework in the frequency domain covering Rayleigh and Rician fading scenarios, with and without branch correlations and/or unequal branch powers. The Rician fading scenario is approximated using a Nakagami-m distribution. The branch correlations and unequal branch powers are dealt with through a simple and novel device, the concatenated array-equivalent channel. Likewise, a novel approach is presented that reduces the Rician fading case to an equivalent Rayleigh-fading system. Furthermore, results are presented for different array topologies where correlation characteristics are obtained with the help of a measurement-validated correlation model situated in the local multipoint distribution service band, i.e., the carrier frequency used in the measurements was 29.5 GHz.

This paper proposes a joint adaptive rate selection and power control algorithm for broadband TDMA/TDM wireless networks. The proposed algorithm is a modified, enhanced, and more robust version of the selective power control with active link protection (SPC-ALP) algorithm proposed for adaptive transmission rate and power control in CDMA networks [R.Jantti and S. Kim, 2000]. Unlike SPC-ALP, the proposed algorithm uses adaptive coding and modulation (instead of variable spreading gain) for transmission rate control. In addition, the proposed algorithm is different in two aspects; the first one is the inclusion of the cochannel interferer assistance mode, while the second one is the use of the signal quality as a criterion for user removal. Results show that the proposed algorithm outperforms SPC-ALP in terms of the net throughput and signal quality measures such as the outage probability and frame error rate. Channel reallocation is also studied and found to be very effective in enhancing the system performance particularly at low to medium loading.

It is well known that link adaptation techniques, when designed to track the channel variations, yield a higher network throughput. In this work, we investigate the throughput returns due to the employment of various combinations of adaptive modulation, adaptive coding, and adaptive power control in a fixed cellular broadband wireless access system incorporating the effects of shadowing, multipath fading, and multiple access interference. The system considered is a multipoint multichannel distribution system (MMDS) with carrier frequency 2.5 GHz. It is observed that among all the possible combinations, the combination of adaptive modulation and adaptive coding (without power control) is the most efficient type since the further employment of adaptive power control only adds a relatively small improvement in the throughput. The frequency of occurrence of different constellation sizes and code rates with the percentages of the successful and failing links, are reported as well in this study.

This paper provides a theoretical characterization of the multihop wireless communications channel without diversity, wherein intermediate terminals relay information without employing spatial diversity techniques. Two channel models are proposed and developed: one where each intermediate terminal digitally decodes and re-encodes the received signal from the immediately preceding terminal and the other where each intermediate terminal simply amplifies the received signal from the immediately preceding terminal. These models are compared, through analysis and simulations, with the single-hop reference channel on the basis of probability of outage and probability of error. Both models achieve significant gains over the single-hop reference channel, with the amplified relaying model matching the performance of the decoded relaying model despite noise propagation.

Digital relaying is a technique that uses certain mobile terminals that have good communication links with the base station to act as relay nodes for those that do not have. With this technique, the signal quality at the destination is expected to improve since the signal only goes through favorable links and, at each intermediate node, the signal is decoded and re-encoded (so that no noise is propagated) before forwarding. One of the major costs of relaying is that additional channels are needed. We investigated the effect of using different schemes for selecting channels - among those already used in the adjacent cells - for relaying on performance improvement. Our simulation results show that not only can relaying improve the performance; but also, with power control this improvement is quite insensitive to the channel selection scheme. Furthermore, with small cell radii where there is a potential for high interference level, only a low relay node transmit power is sufficient to improve the system performance significantly.

An examination of potential advantages of cell-site diversity with selection combining in the context of MM-wave fixed cellular systems is reported. The study involves simulation of converging radio links over weather radar images of the radar reflectivity factor, from which the specific attenuation of rain at 30 GHz is derived. The average correlation of attenuation on two converging links as a function of their angular separation is shown to indicate the potential benefits in the use of cell-site diversity, especially in heavy rain. Results show that diversity gain exhibits a dependence on angular separation /spl theta/ in the general form of sin/sup k/ (/spl theta//2). For links of identical lengths the model reduces to a root-sinusoidal shape (k=0.5), whereas links of unequal lengths lead to the ITU-R recommended model (k=1). Based on the sin/sup k/ (/spl theta//2) model and observation of the length ratio of the links, a set of criteria for determining the benefit of cell-site diversity for a given subscriber location is proposed.

This paper provides a theoretical characterization of the multihop wireless communications channel with diversity, wherein intermediate terminals relay information employing spatial diversity techniques. Two channel models are proposed and developed: one where each intermediate terminal combines, digitally decodes, and re-encodes the received signals from all preceding terminals and the other where each intermediate terminal simply combines and amplifies the received signals from all preceding terminals. These models are compared, through analysis and simulations, with the single-hop reference channel on the basis of probability of outage and probability of error. Both models achieve significant gains over the single-hop reference channel, with the amplified relaying model outperforming the decoded relaying model despite noise propagation.

In this overview, we review the basic reception techniques of adaptive spatial and temporal processing, which have been shown to yield substantial improvements in capacity and performance of wireless systems. After a review of appropriate transmission channel models, we introduce the elements of linear and nonlinear space-time processing and outline their capabilities for combating interference and multipath. We then describe some current applications and discuss adaptation issues.

A new adaptive combined space-time receiver for direct sequence code division multiple access (DS-CDMA) has been proposed previously to combat co-channel interference and enhance the capacity for cellular CDMA systems. It has been shown that this structure improves the cancellation capability of the CDMA systems compared to the conventional CDMA receiver and the adaptive multiuser structure. This structure combines the function of adaptive antenna array, adaptive multiuser detector and the RAKE receiver in one single structure. In this paper we study the transient behavior of the linear space-time receiver using the standard LMS algorithm. Results show that using the linear space-time receiver the training period could be reduced significantly, and the adaptation time is slightly affected by adding more interferers.

The Local Multipoint Communication Services (LMCS) refer to millimeter-wave point-to-multipoint access systems that provide broadband services to both residential and commercial subscribers. Coverage is the most important problem to be resolved for the successful deployment of the LMCS systems. This paper reports the performance study of LMCS in relation to transmit power control. An extensive simulation model has been used to study the effects of a number of factors on the system outage performance. These factors include the propagation model, power control command rate, step size and dynamic range.

Broadband wireless access (BWA) offers attractive features such as ease and speed of deployment, fast realization of revenues, and low infrastructure cost. This paper focuses on medium access control (MAC) alternatives that can find application in an LMDS/LMCS network capable of supporting multimedia traffic. Multifrequency demand assignment TDMA-based schemes appear (at the moment) to be a suitable choice in this context. The selected protocol should be dynamically capable of providing multirate capabilities and quality-of-service guarantees. An obvious candidate for the aforementioned purpose is the LMDS-specific MAC proposal in the DAVIC 1.2 recommendation. For purposes of comparison, we also examine the evolving IEEE 802.14 MAC convergence protocol and MCNS (multimedia cable network system) DOCSIS (ITU J-112) MAC standard that are intended to support similar applications and services in an HFC (hybrid fiber coax) environment. The three protocols are examined under noiseless and noisy channel conditions. Previous results on LMDS channel characteristics are summarized and used for modeling noisy channel conditions. The candidate protocols are compared in terms of mean access delay, throughput, and collision multiplicity statistics, when contention of users is involved. The effect of dynamic slot allocation on the performance of the candidate protocols is also examined.

An adaptive combined array and multiuser detector (spatial and temporal) structure for direct sequence code division multiple access (DS-CDMA) has been proposed in Legnain et al. (1998) to combat co-channel interference and increase the capacity for cellular CDMA. The structure uses a L-sensor (element) linear array. Each element is followed by a chip matched filter (CMF) and a fractionally spaced adaptive equalizer. The proposed structure outperforms the conventional receiver and the structures proposed in Abdulrahman et al. (1994) and Miller (1995) in-terms of capacity and near far resistance. In this paper we propose a new centralized decision feedback space-time receiver, which feeds back the detected symbols for all users through a FIR filter without respreading the detected symbols. The focus in this paper is the co-channel interference rejection capability of this structure. Analysis of the optimum tap weights and the minimum mean square error in multipath (no fading) channels are provided, and comparison between this structure and other structures are also provided.

This paper focuses on medium access control (MAC) alternatives that can find application in a LMDS/LMCS network capable of supporting multimedia traffic. Multi-frequency, demand assignment TDMA-based schemes appear at the moment to be a suitable choice in this context. The selected protocol should be dynamically capable of provision of multirate capabilities and quality of service guarantees. An obvious candidate for the above purpose is the LMDS-specific MAC proposal in the DAVIC 1.2 recommendation. For purposes of comparison, we also examine the evolving IEEE 802.14 MAC convergence protocol that is intended to support similar applications and services in an HFC (hybrid fiber coax) environment. The MCNS (multimedia cable network system) MAC characteristics are also briefly described.

An adaptive combined space-time detector for DS-CDMA system has been proposed by Legnain, Falconer and Sheikh (see IEEE GLOBECOM 98, Sydney, Australia, 1998) to combat cochannel interference and enhance the capacity of the adaptive MMSE multiuser detection. In this paper we study the effect of time varying user population (sudden death and birth) on the performance of the adaptive space-time multiuser detector proposed by Legnain et al. in frequency selective fading channels. Sudden birth of interferers degrades the performance of the adaptive MMSE detector, and the receiver needs to be readapted each time new interferers appear. We show that the adaptive space-time techniques alleviate the effect of sudden birth of interferers and there is no need to retrain the detector each time a new interferer appears or disappears.

Novel algorithms for block equalization of M-ary phase shift keying (PSK) signals transmitted over multipath fading channels in the presence of an interferent cochannel signal are introduced and analyzed. The algorithms exploit the intrinsic statistical properties of cochannel interference (CCI) in order to mitigate its effects. Both linear and decision feedback equalizers (DFEs) are derived under the assumption that the overall channel impulse responses of both the useful and the inteferent signal are known. Simulation results show that: (a) whereas zero-forcing block equalizers yield a large noise enhancement effect, a minimum mean-square block DFE (MMSE-BDFE) can efficiently compensate for the distortion in the useful channel and reduce the effect of CCI at the same time, and (b) the MMSE-BDFEs outperform conventional DFEs, at least in the idealized conditions of our analysis.

This research concerns comparing the performance of various transmission and reception alternatives for the 5-45 MHz portion of the upstream path of the return coaxial cable band. This evaluation is based on ingress noise spectra files collected over an extended period of time from various sites. This data has been used to evaluate the return band capacity as a function of sub-channel bandwidth in FDMA and TDMA/FDMA schemes. In addition, the time varying aspect of the channel is considered using variable bit rate allocation to sub-bands, as in OFDM with frequency division multiplexing. Furthermore, the same analysis is repeated for a wideband system using a decision feedback equalizer (WDFE) contending with ingress noise instead of channel distortion. This research demonstrates that using a variable channel allocation (VCA) scheme to match the time varying aspect of the channel enhances the performance over a fixed channel allocation (FCA) scheme. It also shows that a wideband system using an adaptive decision feedback equalizer (DFE) performs similar to a multi-carrier modulation (MCM) method, and that reasonable performance can be obtained by using a system of lower complexity consisting of the combination of the FCA with the WDFE.

With the development of wireless and personal communication systems, there is considerable interest in using antenna array technology to improve the system capacity in both present and future generation wireless systems. A novel and simple algorithm is proposed for blind adaptive extraction of a binary phase-shift keying (BPSK) signal in the presence of interference by cyclostationary signal processing using an antenna array. The algorithm operates in an interference-limited system in which the desired and interfering signals have identical symbol rates, but are modulated on slightly different carrier frequencies. Compared to existing blind algorithms which also exploit the cyclostationarity of the received signal, the new algorithm provides a simpler and faster converging means to estimate the channel phase for diversity combining. Analytical and simulation results are presented and performances are compared with direct matrix inversion (DMI) and existing blind algorithms. Based on the proposed algorithm and the result obtained by Gardner, Schell and Murphy (1992), a time-division multiple-access (TDMA) system with the new channel assignment scheme called cyclic TDMA (CTDMA) is proposed. The simulation results also show that the proposed algorithm is relatively simple and very promising in applications to indoor wireless communication where interference rejection and increased spectrum efficiency are the objectives. Analysis and simulation results are presented to confirm the interference rejection capabilities. The robustness of existing and proposed algorithms to the perturbation of cyclostationarity is also discussed.

Antenna diversity is an important technique to combat fading and reduce co-channel interference (CCI). When evaluating the system performance, such as the BER, Monte Carlo simulation is widely used due to the difficulty of analysis. This paper presents the analysis of BER for optimum combining (OC) with two cochannel interferers and maximal ratio combining (MRC) with arbitrary number of interferers. For a DPSK signal, an exact BER expression is derived. The paper also provides an upper bound for BPSK or QAM signals based on results of Foschini and Salz (1983).

The decorrelating detector is a near-far resistant linear joint detector for a code-division multiple-access (CDMA) system. It consists of a bank of matched filters followed by a decorrelating matrix. For proper operation, both the matched filter bank and the decorrelating matrix require knowledge of the delays of all users. The delays are due to the different propagation times from each user to the base station. Delay estimation is a weak link in this system since it is complex and prone to error. The proposed decorrelating detector does not require exact knowledge of the user delays, but instead requires that the delays be bounded to a fraction of a symbol interval. The delays are naturally bounded in this way in many microcell and picocell systems where the round trip propagation time is small compared to the symbol interval. The new delay independent decorrelating detector is shown to be near-far resistant and, through appropriate spreading code selection, to experience a modest 3 dB signal-to-noise ratio (SNR) loss relative to orthogonal access schemes. It is also shown to limit capacity to a maximum of 50% of the spreading gain when the delays are bounded by a single chip interval. The complexity is similar to the conventional correlating receiver which is far less complex than the joint detection schemes proposed to date.

An adaptive polling scheme for multiple access in a broadband wireless local area networks (LANs) is proposed. It is designed to accommodate asynchronous transfer mode (ATM) traffic with a large and variable range of bit rates. The radio architecture is microcellular, with switched sectored-beam antennas and a small frequency reuse factor. Features of the multiple access scheme include polling rate depending on a terminal's average bit rate, fixed-length frames divided into segments corresponding to base antenna sectors, and automatic-repeat-request (ARQ) for error control, combined with randomized sector segment ordering. The steady-state performance (cell loss, call blocking and dropping probabilities, waiting time, and buffer occupancy) are evaluated by simulation for an open indoor environment, and for handoff, ARQ, and simple power control enhancements; focusing mainly on the effects of the propagation and interference conditions.

Adaptive equalization at the mobile in conjunction with diversity transmission at the base station can combat the adverse effects of both frequency selective and frequency non-selective fading at the mobile receiver. This is suggested as an alternative to diversity combining at the mobile. Thus a substantial saving in the mobile receiver's implementation cost and power consumption is expected. A North American digital cellular radio standard TDMA system with two transmitting antennas at the base station and MLSE at the mobile is considered. System performance results in the presence of severe Doppler frequency shifts and additive white Gaussian noise in a flat fading channel are presented.

In a typical indoor channel, the channel dispersion varies from 10 ns to 100 ns depending on the physical dimensions of the building, the material of the walls etc. Therefore, for high rate data transmission around 160 Mbps, the signals experience severe intersymbol interference (ISI). A simple channel equalization strategy is described for data transmission at around 160 Mbps in the millimeter wave frequency band. CPFSK, a constant envelope modulation, is selected as the modulation scheme. The proposed receivers consist of a noncoherent demodulator followed by a nonlinear equalizer. Performance results of the proposed systems are presented for a two path quasi-static channel.

An adaptive polling scheme for multiple access in a broadband wireless local area networks (LANs) is proposed. It is designed to accommodate asynchronous transfer mode (ATM) traffic with a large and variable range of bit rates. The radio architecture is microcellular, with switched sectored-beam antennas and a small frequency reuse factor. Features of the multiple access scheme include polling rate depending on a terminal's average bit rate, fixed-length frames divided into segments corresponding to base antenna sectors, and automatic-repeat-request (ARQ) for error control, combined with randomized sector segment ordering. The steady-state performance (cell loss, call blocking and dropping probabilities, waiting time, and buffer occupancy) are evaluated by simulation for an open indoor environment, and for handoff, ARQ, and simple power control enhancements; focusing mainly on the effects of the propagation and interference conditions.

For CDMA (code-division multiple-access) wireless communications systems, orthogonal signaling with noncoherent detection has been proposed when a reference phase is not available. In this paper, orthogonal signaling using Walsh signals is considered. Performance of the channel using error-correcting coding together with 1) bit-by-bit interleaving or 2) Walsh word-by-word interleaving is investigated. Simulation results show that for several types of fading and non-fading channels with practical amounts of interleaving and with practical power control, the second technique with complete soft decision decoding that has added complexity in the decoder gives significant reduction in the required signal to interference ratio (SIR) over the first technique.>

The paper concerns the feasibility and achievable performance of adaptive filtering in an interference-limited multipath fading environment as encountered in indoor wireless communications. In a typical cellular radio application, the performance-limiting impairment is interference due to synchronous data streams from other co-channel and adjacent channel users (CCI and ACI). The receiver under consideration employs an adaptive fractionally spaced decision feedback equalizer (DFE) Which exploits the correlation of the cyclostationary interference to achieve superior performance relative to the worst case when the interference is stationary noise. The paper presents ideal calculations which confirm that significant DFE performance gains are potentially achievable by explicitly accounting for the cyclostationary CCI. Two adaptive DFE strategies are considered. One approach is to adapt the DFE directly using iterative algorithms such as least mean square (LMS) or recursive least squares (RLS). Another approach is to compute the minimum mean square error DFE using an RLS channel impulse response (CIR) estimate and a sample estimate of the CCI autocorrelation obtained from the CIR estimation error during training. The best approach for adaptive equalization, in terms of adaptation speed and system performance, is to employ an RLS DFE which does not explicitly estimate the CIR or the CCI autocorrelation.>

Antenna diversity transmission methods are used to improve the performance of wireless systems. Delayed diversity transmission (DDT) is an attractive option when the main issue is complexity versus performance. The paper discusses the application of delayed diversity transmission to indoor wireless systems in order to reduce the implementation complexity of both base stations and portable terminals. System configurations are suggested that improve the system performance in both the forward and reverse direction. Through computer simulations, the optimum delay, which is the key parameter for the system performance, is derived for an arbitrary frame format. Subsequently, the system performance is tested for the IS-54 standard format structure. DDT shows promise in the reduction of both the power consumption and implementation costs of indoor wireless systems.

Investigates the average bit error rate (EER) and outage performance of a TDMA indoor wireless cellular system employing an adaptive fractionally spaced decision feedback equalizer (DFE) in the presence of adjacent and co-channel interference (ACI and CCI) and additive Gaussian noise. There exist performance gains in the presence of strong interference and negligible ambient noise relative to the equivalent stationary noise case. Moreover, a directly adapted RLS DFE performs better than a computed MMSE DFE which employs estimates of the channel impulse response (CIR) and interference plus noise autocorrelation. However, these comparative performance gains are compromised by the presence of appreciable noise power and/or more interferers than the receiver can process effectively, The use of a wide receiver bandwidth yields a performance improvement for channel spacings which allow for sufficient spectral overlap of the ACI with the desired signal bandwidth. Thus, a reduction in channel spacing increases the radio capacity while maintaining a desired average EER or outage performance.>

Commercial interest in Code Division Multiple Access (CDMA) systems has risen dramatically in the last few years. It yields a potential increase in capacity over other access schemes, because it provides protection against interference, multipath, fading, and jamming. Recently, several interference cancellation schemes for CDMA have been proposed but they require information about all interfering active users or some channel parameters. The authors present an adaptive fractionally spaced decision feedback equalizer (DFE) for a CDMA system in an indoor wireless Rayleigh fading environment. This system only uses information about the desired user's spreading code and a training sequence. An analysis on the optimum performance of the DFE receiver shows the advantages of this system over others in terms of capacity improvements. A simulation of this system is also presented to study the convergence properties and implementation considerations of the DFE receiver. Effects on the performance because of sudden birth and death of users in the CDMA system and bit error rate performance of the DFE receiver is also presented.>

CDMA (code division multiple access) may be an attractive technique for wireless access to broadband services because of its multiple access simplicity and other appealing features. In order to investigate the traffic handling capabilities of a future network providing a variety of integrated services, this paper presents a study of a broadband indoor wireless network supporting high-speed traffic using CDMA multiple access. The results are obtained through the simulation of an indoor environment and the traffic capabilities of the wireless access to broadband 155.5 MHz ATM-SONET networks using the mm-wave band. A distributed system architecture is employed and the system performance is measured in terms of call blocking probability and dropping probability, the impacts of the base station density, traffic load, average holding time, and variable traffic sources on the system performance are examined. The improvement of system performance by implementing various techniques such as handoff, admission control, power control and sectorization are also investigated.>

In order to make the analytical evaluation of portable radio link performance tractable while avoiding lengthy simulations, intersymbol interference and cochannel interference are often modeled as non-fading, Gaussian noise. This paper investigates the viability of such an approach for wideband channels where equalization and space diversity are used to combat interference and signal fading. Comparisons of results obtained using such an assumption of interference with those of a simulation-based approach indicate that as long as the number of cochannel interferers is smaller than the space diversity order, the variance of the Gaussian interfering symbols can be adjusted to provide a reasonable estimate of performance over error rates of interest.>

We describe the ability of a linear equalizer/combiner or decision-feedback equalizer to suppress all received adjacent-channel, intersymbol, and cochannel interference. The emphasis is on values among transmitter bandwidth, receiver bandwidth, carrier spacing, and antenna diversity which provide the best opportunities for interference suppression. Through analyses of the number of degrees of freedom and constraints in generalized zero-forcing equalizers, and partial comparisons to calculations of equalizer minimum-mean-square performance, four results are obtained. First, with one antenna and a linear equalizer, arbitrarily large receiver bandwidths allow for marginal improvements in spectral efficiency through decreased carrier spacing, because the carrier spacing cannot be reduced to a value below the symbol rate without incurring unsuppressible interference. Second, large receiver bandwidths assist multiple antennas in improving the spectral efficiency in that carrier spacing values may go below the symbol rate, even in the presence of cochannel interference. Third, the use of equalizers and linear combiners, together with large receiver bandwidths, allows large transmitter bandwidths to be used. Fourth, for cochannel interference and intersymbol interference, the number of interferers that may be suppressible by a generalized zero-forcing linear equalizer/combiner increases linearly with the product of the number of antennas and the truncated integer ratio of the total bandwidth to the symbol rate.>

Personal communications is a rapidly expanding field with many potential applications. The wireless communication systems designed for indoor environment are expected to handle voice and data services. The millimeter wave band appears to be a favorable choice as it meets the requirements of sufficient bandwidth, sparse usage, small RF device dimensions and favorable regulatory environment. It also has the potential to support broadband service access. This paper reports the results of a computer simulation tool developed for evaluating the traffic handling capabilities of time division multiple access (TDMA) system operating with time division duplex (TDD) in indoor environment. The simulator is used to quantitatively present the impact of blocking probability, traffic offered by the portable, portable density and call congestion on system performance.>

Interference from digital signals in multipair cables has been shown to be cyclostationary under some conditions. This work evaluates the performance of a decision feedback equalizer (DFE) in the presence of cyclostationary interference (CI), intersymbol interference (ISI), and additive white noise (AWN). A comparison between a DFE with CI and one with stationary interference (SI) shows the ability of the DFE to substantially suppress CI. Fractionally spaced and symbol-rate DFE equalizers are also compared and the former is found to yield better performance, especially in the presence of CI. The use of a symbol-rate DFE using an adaptive timing technique that finds the receiver's best sampling phase is proposed for when the fractionally spaced DFE cannot be used because of its complexity. The results also demonstrate the potential benefits of synchronizing central office transmitter clocks, if a fractionally spaced DFE is used at the receiver.>

Compound strategies of equalization and space diversity in the form of an optimum baseband combiner are attractive for wide-band time division multiple access (TDMA) portable communication radio links in order to combat dispersive fading and cochannel interference. This paper investigates the performance of such a scheme in conjunction with convolutional coding and soft-decision Viterbi decoding via a semi-analytical technique based on the method of moments. Such an approach avoids a Gaussian characterization of interference and yields results for both ideal interleaving and no interleaving. With dual space diversity, three taps per forward filter and a data rate of 10 Mb/s, it is shown that, although a third space diversity branch remains preferable in terms of performance, channel coding can be a viable alternative, particularly in terms of outage rate, to increasing the space diversity order even in the absence of interleaving provided the signal-to-interference ratio is sufficiently high

The authors present a simulation study evaluating the performance of trellis-coded modulation with combined code/ISI sequence estimation for high bit rate (800 kb/s) transmission on subscriber loops. The receiver contains a fractionally spaced forward filter of a decision feedback equalizer (DFE) as a front end. This is shown to suppress phase synchronized crosstalk very effectively. The performance is further enhanced by the use of trellis code with a large number of states received with an M algorithm sequence-estimating receiver with a smaller number of states.>

Equalization in cyclostationary interference, which occurs when similar neighboring digital communication systems cause interference (crosstalk), is considered. An expression is derived on the minimum mean square performance of a continuous-time infinite-length decision feedback equalizer in the presence of multiple cyclostationary interferers and additive white noise. This expression is calculated for a subscriber line system to show the performance improvements over the situation where the interference is stationary with the same power spectrum. Linear equalizer performance curves were also added to the comparisons. These results show two techniques which can provide opportunities for improved equalizer performance by enhancing the cyclostationarity of the interference. The first is by decreasing the misalignment of the phases of the transceiver clocks in the central office transmitters. The second is by using transmitter pulse bandwidths which are wide relative to the symbol rate.>

A system model is proposed that allows one to apply both trellis coding and PN spreading sequence to the data symbols to be transmitted. Rate n/n+1, trellis codes using 2/sup n+1/-point MPSK signal constellations are investigated when Gold sequences are used for PN spreading. Performance in an additive white Gaussian noise (AWGN) channel is investigated, with 5-20 users transmitting simultaneously. Using the criteria of equal complexity and throughput, the performance of the trellis codes in a SSMA (spread spectrum multiple access) environment is compared to that of medium-rate to low-rate convolutional codes through the use of a generalized transfer function bound. The average degradation due to the interuser interference is determined by the method of moments. The validity of approximating the interuser interference as a Gaussian random variable is also investigated. The numerical results illustrate that for a given complexity, chip rate and throughput, low-rate convolutional codes provide the best performance in an SSMA system. As lower-rate convolutional codes are used, there is an increase in the effective interuser interference due to the greater cross-correlation effects from using shorter PN sequences, or alternatively from the effects of partial cross-correlation. However, this increased degradation is more than overcome by the increased distance properties of the low-rate codes.>

The use of reduced-state sequence estimation techniques in a digital subscriber loop receiver is discussed. These techniques offer a potential performance improvement over conventional equalization techniques such as decision feedback equalization (DFE). Stationary and cyclostationary NEXT noise models are described. The theoretical performance obtainable from a Viterbi algorithm receiver with stationary white Gaussian noise, stationary NEXT, and cyclostationary NEXT noise models is estimated, and the reduced-state decision feedback sequence estimation and M algorithms are reviewed. It is shown that the improvement can be especially significant in the presence of cyclostationary crosstalk because of the freedom that sequence estimation receivers afford in the choice of receiver sampling phase. This advantage is evaluated for Viterbi algorithm receivers. By simulation of two practical reduced-state sequence estimation receivers, it is demonstrated that, in the presence of cyclostationary crosstalk, a substantial increase in maximum loop range (or equivalently, maximum bit rate) may be achievable compared to conventional DFE equalization.>

Data transmission at rates of 1.2 kbits/s or higher through voiceband ionospheric channels is subject to impairment from severe linear distortion, fast channel time variations, and severe fading. In this paper, we have focused on the performance of DFE (decision feedback equalization) receivers for communication over 3 kHz bandwidth HF channels. We describe the results of simulations for a wide range of fading rates on simulated and real recorded HF channels, using fractionally spaced DFE receivers. Both LMS (least mean square) and FRLS (fast recursive least squares) adaptation algorithms with periodic restart were evaluated, and both ideal-reference and decision-directed operation was observed. The results indicate that FRLS adaptation yields superior performance to LMS in rapid fading conditions, but that this performance advantage diminishes at low signal-to-noise ratios. Also, fade rates greater than about 1 Hz produced relatively high error rates, irrespective of which adaptation method was employed. Finally, a novel modification of the simple LMS algorithm which improves its tracking ability was evaluated. This involved preceding the LMS DFE receiver with an adaptive lattice whitening filter.

Subjective quality measurements on three digital speech coders, simulated with mobile radio channel transmission, were performed using the "mean opinion score (MOS)" method. The three speech coding methods tested were: continuously variable slope deltamodulation (CVSD) coding, adaptive predictive coding (APC), and residually excited linear predictive (RELP) coding. Several versions of each coder, with transmission rates in the range of 7.3 to 16.1 kbits/s, were simulated. Five different channel conditions, including three derived from land mobile radio field experiments, were applied to the speech coders' encoded output to study the effects. The results show that of the three coders, the CVSD coder is the most robust to channel errors, but produces reconstructed output speech of unacceptable quality. The 14.4 kbit/s RELP coder produces relatively good Output speech quality, exhibits a mild degree of robustness to mobile radio channel errors, and is slightly less complex than the APC coder. Of the three digital speech coders tested, the RELP coder appears the most suitable for use with land mobile radio. However none of the three coders was able to produce speech of telephone toll quality in a mobile radio environment.

The degree of echo reduction achieved in a digital subscriber loop (DSL) transceiver employing adaptive echo cancellation has previously been shown to be very sensitive to timing jitter. This paper reports the experimental study, by computer simulation, of a 144 kbit/s DSL timing recovery system incorporating a prefilter, nonlinearity, and second-order phase-locked loop. The simulations were carried out for a wide variety of line lengths, bridged-tap conditions, loop and filter parameters, and choice of nonlinearity. The results showed that a properly designed timing recovery system with a 20 khzwide prefilter allows satisfactory echo reduction performance, even for long loops with bridged taps. For shorter loops the achievable echo reduction is less, but is still sufficient to yield satisfactory performance.

This paper treats analytically and experimentally the response of RLS {Recursive Least Squares} adaptive filters with exponential windows to stationary and nonstationary inputs. A new formula for the "estimation-noise" has been derived involving second- and fourth-order statistics of the filter input as well as the exponential windowing factor and filter length. Under general time-varying conditions it is shown that the time constant associated with "lag effects" depends solely on the exponential weighting parameter λ. In addition the calculation of the excess mean square error due to the lag for an assumed Markov channel provides the necessary information about tradeoffs between speed of adaptation and steady-state error. In the simple case of channel identification it is shown that the LMS and RLS adaptive filters have the same tracking behavior.

We analyze the effect of Subscriber-end timing recovery Circuit jitter on the performance of two types of adaptive echo cancellers that can be used for full-duplex digital transmission on two-wire subscriber loops. Under severe echo-to-far-end signal ratios, echo canceller performance is found to be quite sensitive to high-frequency jitter components. Satisfactory performance with respect to jitter requires that a narrow-band phase-locked loop, rather than a single-tuned high-Qfilter, be employed for timing recovery.

The performance of an adaptive echo canceller at the looptimed subscriber end of a digital subscriber loop has been shown to be sensitive to jitter arising from the timing recovery subsystem. We evaluate this degradation for the common timing recovery subsystem consisting of a prefilter, a squarer, and a second-order phase-locked loop. The evaluation shows the influence of equalization, prefilter shape, and phaselocked loop parameters. A narrow-band accurately tuned prefilter, line equalizer, and a narrow-band phase-locked loop are found to be necessary for adequate performance in a 144 kbit/s bipolar-coded digital subscriber loop system employing echo cancellation.

Data communication at rates near or above 2 kbits/s on 3 kHz-baadwidth HF radio channels is subject to impairment from severe linear dispersion, rapid channel time variation, and severe fading. In this investigation, recorded 2.4 kbit/s QPSK signals received from HF channels were processed to extract a time-varying estimate of the channel impulse response. From the estimated channel impulse responses, performance-related parameters were computed for ideal matched filter reception, maximum-likelihood sequence-estimation (MLSE), and decision feedback equalization (DFE). The results indicated that the simpler DFE receiver suffered only a small theoretical performance degradation relative to the more complex MLSE receiver. Other HF channel impulse response statistics were also obtained to shed light on equalization and filter adaptation techniques.

The AREC (adaptive reference echo cancellation) algorithm is presented for an echo canceler used in full-duplex two-wire digital transmission on digital subscriber loops. The AREC algorithm incorporates a decision-directed estimation of and compensation for the far-end signal which is a source of interference to the conventional echo canceler adaptation algorithm. The AREC algorithm thus offers much faster convergence and shorter coefficient wordlengths than the conventional algorithm. Analysis and simulation of the performance and convergence of both AREC and conventional echo canceler adaptation algorithms are carried out. Included in the analysis is the effect of receiver delay and coefficient wordlength requirements. A simple and robust startup procedure is proposed and investigated by simulation.

Very rapid initial convergence of the equalizer tap coefficients is a requirement of many data communication systems which employ adaptive equalizers to minimize intersymbol interference. As shown in recent papers by Godard, and by Gitlin and Magee, a recursive least squares estimation algorithm, which is a special case of the Kalman estimation algorithm, is applicable to the estimation of the optimal (minimum MSE) set of tap coefficients. It was furthermore shown to yield much faster equalizer convergence than that achieved by the simple estimated gradient algorithm, especially for severely distorted channels. We show how certain "fast recursive estimation" techniques, originally introduced by Morf and Ljung, can be adapted to the equalizer adjustment problem, resulting in the same fast convergence as the conventional Kalman implementation, but with far fewer operations per iteration (proportional to the number of equalizer taps, rather than the square of the number of equalizer taps). These fast algorithms, applicable to both linear and decision feedback equalizers, exploit a certain shift-invariance property of successive equalizer contents. The rapid convergence properties of the "fast Kalman" adaptation algorithm are confirmed by simulation.

Asymptotically optimum (in the sense of minimum per-symbol error rate) receiver structures for data communication over the white Gaussian channel with unknown time delay and carrier phase jitter are developed. The receiver structures apply to the following suppressed-carrier modulation systems: double sideband (DSB), quadrature amplitude modulation (QAM) with an arbitrary constellation, vestigial sideband (VSB) and single sideband. The resulting minimum error probability receivers are asymptotically equivalent to maximum-likelihood digital {\em sequence}-estimating receivers. The optimum structures implicitly derive joint maximum-likelihood estimates of the unknown parameters and of the sequence of data symbols. It is shown that the parameter estimates can be obtained from two data-directed stochastic approximation algorithms. Unlike traditional theoretical treatments of this communication situation, which have separated the highly important carrier phase and timing recovery problem from the detection problem, a unified theory is presented from which the complete ideal receiver structure can be deduced.

We describe and experimentally evaluate a receiver structure for two-dimensional data communication systems based on passband decision feedback equalization and data-directed carrier recovery. The forward and feedback tap coefficients and the demodulator's carrier phase angle are adjusted adaptively; the carrier recovery and demodulation operation, as in the case of a linear equalization receiver, can compensate for rather severe frequency offset and phase jitter. Experimental results are reported of simulations of the receiver on voiceband telephone channels.

The application of QAM receivers employing passband decision feedback equalization or Viterbi algorithm detection is considered for high-speed data transmission in voiceband telephone channels. Analytical results for the Viterbi receiver indicate that it can permit digital data transmission with symbol rates exceeding the nominal bandwidth capabilities of the channel. However, it is also shown that the minimum distance of partial response-type impulse responses can substantially decrease due to small demodulation phase errors. Thus, decision feedback receivers may offer more robust performance on channels with phase jitter.

This paper describes experimental results of simulated 12000 and 14400 bits/s quadrature amplitude-modulated (QAM) systems (which were described in detail in Part I [1] ) operating on real and simulated voiceband channels. These experimental results show that both Viterbi algorithm detection and decision feedback equalization allow digital data transmission at symbol rates exceeding the nominal bandwidth capabilities of typical telephone channels. It is demonstrated in these experiments that the Viterbi algorithm receiver structure tested is less affected by linear distortion than the decision feedback receiver but exhibits greater sensitivity to phase jitter. On conditioned voiceband channels where there may be substantial phase jitter, a reasonable compromise between sensitivity to distortion and noise and sensitivity to phase jitter is made by a modem employing decision feedback equalization and decision-directed carrier phase tracking.

An upper bound is obtained on the probability that a specified pattern of errors, caused by a combination of intersymbol interference and additive Gaussian noise, occurs in an amplitudemodulated digital data transmission system. The bound, which is a generalization of Saltzberg's upper bound for the probability of a single error, behaves as an exponential function of the signal-to-noise ratio and the overall channel impulse response. The exponent of the bound for a pair of errors is calculated for two "typical" channels and in each case is fairly close to twice the corresponding Saltzberg exponent for a single error.

We consider a suboptimum decision-directed block decoder for a binary symmetric channel that makes use of past decoding decisions to update its estimate of the channel's initially unknown crossover probability. The decoder has a threshold list decoding rule that uses the current estimated crossover probability. The estimate is updated by means of a stochastic approximation algorithm. It is shown to converge toward the true crossover probability with a bias that decreases exponentially with the code's block length, provided it never "runs away" toward zero after dropping below a certain critical value. The probability that this runaway phenomenon ever occurs is bounded by an expression that is exponentially decreasing in the code's block length and in the weight assigned to the initial estimate.