In this paper, successive intracell interference cancellation (IIC) of the wideband-code division multiple access (W-CDMA) signal at the mobile unit is considered. Three new interference cancellation techniques suitable for the downlink of any CDMA system with orthogonal spreading are proposed. No prior knowledge of users' spreading codes or even their spreading factors are required for interference cancellation. A new term, effective spreading code, has been introduced, which is defined as the interfering user physical code as seen by the desired user within the desired user symbol duration. The mobile receiver estimates the effective spreading codes of the interfering users regardless of their spreading factors using fast Walsh transform (FWT) correlators (instead of the regular correlators) and uses this information to suppress the intracell multiuser interference. Three different interference-suppressing techniques are studied: subtraction; combined interfering signal projection; and separate interfering signal subspace projection. The complexity of the proposed techniques is low compared to conventional interference cancellation techniques. For a W-CDMA system and the IMT-2000 vehicular channel model, a capacity increase of up to 150% of the original (without IIC) system capacity is shown.

We analyze the performance of a flexible multirate scheme for direct-sequence code division multiple-access (CDMA) mobile radio systems. The proposed scheme uses a variable processing gain serial pseudonoise modulation as a multirate strategy. To reduce the interference effects, the CDMA system utilizes the coherent fast Walsh transform transmission technique. The proposed scheme can be used in the reverse link (mobile-to-base station) of the upcoming third-generation wide-band CDMA system (has the feature of coherent reverse link). We analyze the system performance with and without using a decorrelating multiuser detector. The uncoded bit-error probability (BEP) with and without decorrelating detection on a multipath fading channel is derived analytically. In addition, the validity of the analysis results is demonstrated by computer simulations using the IMT-2000 vehicular multipath channel model. In order to make sure that the proposed processing techniques do not distort the soft values at the demodulator output, the proposed multirate scheme is also simulated in case of using turbo codes. The turbo-coded BEP is calculated for different user data rates and different number of decoding iterations.

The capacity and signal quality of wireless communication systems can be improved dramatically by enhancing the receiver performance in the presence of adjacent channel interference (ACI). The sequence error rate is minimized through joint demodulation of all signals, but with high implementation complexity. Complexity can be greatly reduced through the use of successive cancellation, in which signals are detected and subtracted in descending order of signal power. However, successive cancellation performs poorly when signals have similar power levels. In this paper, an improved successive cancellation receiver for adjacent channel signals is proposed. The conventional detector is replaced with a simple form of joint demodulation. As a result, the receiver combines the benefits of joint demodulation, which works well when signals have similar power levels, with the benefits of successive cancellation, which works well when signals have dissimilar power levels. Performance results for the Global System for Mobile (GSM) communication digital cellular system show an order of magnitude reduction in bit-error-rate when adjacent channel signals are closely spaced and have comparable power levels.

Turbo-codes have drawn much attention due to its remarkable error correcting capability in digital communication systems for long block sizes. The performance of turbo codes for short block sizes suitable for real time communication systems has gained a lot of attention. We consider the applicability of such codes to a WCDMA radio link based on the requirements of the wireless third generation (3G) systems. We present unequal rate turbo codes (URTC) as an alternative to the conventional equal rate turbo code (ERTC) to obtain an overall code rate equal to 1/4. The URTC provides a reduction in the decoding complexity by almost 33%. This is because the unequal rate turbo decoder uses two soft output Viterbi algorithm (SOVA) decoders while the equal rate turbo decoder uses three SOVA decoders. The performance of the URTC, conventional ERTC, and a convolutional code with the same decoding complexity are considered in AWGN as well as flat fading channels.

In this paper, blind intra-cell interference cancellation (IIC) of the W-CDMA signal at the mobile unit is considered. Two new blind interference cancellation techniques suitable for the downlink of any CDMA system are proposed. The receivers are blind in the sense that no prior knowledge of users' spreading codes or even their spreading factors are required. The mobile receiver estimates the effective spreading codes of the interfering users regardless of their spreading factors using fast Walsh transform (FWT) correlators and uses these information to cancel out the intra-cell multi-user interference. The complexity of the proposed techniques is low compared to conventional interference cancellation techniques. Simulation results show a capacity increase of up to 50 % of the original (without IIC) system capacity or up to 10 dB signal to Gaussian noise ratio gain at a fixed capacity (the inter-cell interference is modeled as a Gaussian noise).

In this paper we describe a temporal smoothing approach for estimating the direction of arrival of coherent signals in a fading channel. Over a period of several frames, the multipath signal phases in a fading channel give rise to randomly varying phases. By temporally averaging over these phase terms in a noise subspace estimator such as MUSIC, the received signal array covariance matrix will become full rank, even in the presence of coherent signals. Theoretical results demonstrate that, given K directions of arrival, temporal smoothing requires K+1 sensor elements compared with spatial smoothing which requires 2K elements, and forward/backward spatial smoothing which requires 3K/2 sensor elements.

The capacity and signal duality of wireless communications systems can be improved by enhancing receiver performance in the presence of adjacent channel interference (ACI). In this paper, an MLSE receiver for joint demodulation of adjacent channel signals is developed. This receiver provides significant improvement in performance over a wide range of carrier to interference ratios. A practical approach is taken, in which conventional down-conversion of each signal is performed followed by joint demodulation at baseband. To provide coherent reception, a novel form of joint channel estimation is used, exploiting knowledge of the transmit and receive filter responses. Receiver performance is evaluated via simulation for the GSM digital cellular system.

We derive expressions for the correlation of fading between a pair of antenna elements in a wireless communication system. Some prior researchers have assumed uniformly distributed signal angles of arrival with a specified beamwidth. Here, we derive a closed form expression for the angle of arrival density at a cellular base station where local scatterers are uniformly distributed in a circle about the mobile. We show that the density can be parameterized in terms of the ratio of the base-to-mobile distance to the scattering circle radius, D/R. We use this density to derive the correlation of fading between a pair of antenna elements. We then derive a closed form approximation for the correlation of fading and show that the approximation improves with increasing D/R. We show that if the signal arrives broadside to the antenna elements, then the approximation is exact.

Orthogonal frequency division multiplexing (OFDM) is being considered for mobile communications. This scheme has proven to be very efficient in high communications and for communications over dispersive channels. One of the main obstacles against OFDM is its signal envelope fluctuations. In previous work, a reduced mobile complexity scheme for OFDM has been proposed and analyzed. The envelope fluctuations have been studied and analyzed for a special case of unlimited gain correction. In this paper, a generalized analysis on envelope fluctuations is performed. This analysis includes the effect of varying the gain correction limit on envelope fluctuations in order to find its optimum value.

We simulate a flexible direct sequence-code division multiple access (DS-CDMA) multi-rate system that uses a serial variable processing gain PN modulation together with multi-code modulation. The advantage of serial transmission is that only one waveform is transmitted at a given time resulting in a constant envelope transmission. Multi-code modulation provides a reliable high-speed data communications by using the fast Walsh transform (FWT). The performance of the proposed system is evaluated using computer simulation of different groups of data rates through a frequency selective Rayleigh multipath fading channel.

This paper introduces an adaptive cascaded combination of cancelling co-channel interference (CCI) and decoding of error-correcting codes for a DS-SS CDMA mobile communication system. The analysis of a previous study by Shaheen and Gupta (see IEEE IPCCC Proceedings, Phoenix, Arizona, March 1995) indicated that conventional CCI cancellation techniques might not be as effective, especially for heavily loaded mobile communication systems. This paper extends this work by incorporating integrated filtering and decoding techniques. The extended system is evaluated under the same extreme interference conditions where the receiver is subjected to intersymbol interference (ISI), multi-access interference (MAI), and co-code CCI (CCCCI) generated within the mother cell and from the neighboring cells. Theoretical and simulation results are developed to evaluate the proposed system performance using signal to noise ratio (SNR) and bit error rate (BER) versus number of interfering users from within the cell and from the surrounding cells, and the normalized mean square error (MSE) in parameters estimation.

Uplink cochannel interference between minimum-shift-keyed (MSK) time-division multiple access (TDMA) data packets generated by transmitters distributed uniformly over a circular area has been simulated in software. Distance-dependent attenuation, log-normal shadowing and multipath fading have been implemented. After a discussion of the radio link model and an analysis of bit error probability versus signal-to-interference ratio of a link with a single interferer, a computer model to simulate multiple interferers is presented. Numerical bit error rate (BER) results obtained from computer simulations of a TDMA radio link subject to interference front simultaneous independent transmissions, attenuation, shadowing and fading are compared with known theoretical results for the MSK channel under additive white Gaussian noise (AWGN) only. Possible improvements and future applications of the model to the problem of resource sharing between coexisting networks are suggested.>

This work presents the performance of the direct-sequence spread-spectrum (DS-SS) parallel acquisition system, previously proposed by the authors (1989, 1990), for nonselective and frequency-selective Rician (i.e. specular plus Rayleigh) fading channels. The acquisition system utilizes a bank of parallel I-Q noncoherent matched filters for the search mode, and a coincidence detector for the verification mode. The probabilities of detection and false alarm are derived and the mean and variance of the acquisition time are evaluated as a measure of the system performance. The nonselective channel is a Rayleigh fast fading channel, while the frequency-selective channel model is the so-called wide sense stationary uncorrelated scattering (WSSUS), selective only on frequency. These channels are typical for aircraft-satellite and line-of-sight (LOS) communications.>

A parallel acquisition scheme for direct-sequence spread-spectrum systems is proposed, and its mean acquisition time performance is analyzed in both nonfading and Rayleigh-fading environments. An expression for the mean acquisition time is derived in terms of the probabilities of detection, missing, and false alarm. These parameters are first analyzed for a typical AWGN (additive white Gaussian noise) channel, then the Rayleigh-fading channel encountered in a typical UHF or microwave land mobile radio channel is studied. The channel is assumed frequency nonselective, and the effect of data modulation and code Doppler is not considered. The performance of the parallel system is compared to the corresponding serial system, and it is shown that a significant improvement can be achieved.>

The theory and design of systolic arrays for Viterbi processing in communication systems with a time-dispersive time-varying channel is discussed. The architecture, algorithms, and processor elements, for a two-dimensional systolic array are described. The array supports the branch metric computations required for an adaptive Viterbi processor. The array is designed so that computations propagate along the rows of the array, while data symbols propagate along the columns. All interprocessor data flow and connections within the array are nearest-neighbor. The array illustrates how the Viterbi-processor algorithms can be structured to achieve a high degree of computational concurrency. Variations in the array design are described and evaluated in terms of computational resource requirements and utilization and computational throughput. A high-bandwidth memory interface is proposed, and system design considerations are discussed.>

Partial response continuous phase modulation (CPM) schemes have found wide acceptance because of their compact spectra and comparable performance with other traditional modulation schemes. Although optimum receivers are complex, simple suboptimum receivers are found to yield very good performance in special cases. Performance of such modulation schemes is of interest in a multiple user environment where adjacent channels are spaced closely to improve the system capacity. This paper presents the performance of partial response CPM in the presence of adjacent channel interference and Gaussian noise. The meansquare crosstalk in CPM systems employing MSK-type receivers is formulated. Based on this formulation, a number of modulation schemes employing different receiver filters are analyzed for their ACI rejection. Comparison of results proves that receiver filters, in addition to the spectral occupancy of the signal, play an important role in deciding the crosstalk. The error performance of various schemes evaluated using simulation technique is compared, and it is found that in the presence of adjacent channel interference, certain schemes perform better than minimum shift keying (MSK). The results of the simulation further prove the inadequacy of Gaussian assumption for the adjacent channel interference. It is seen that judicious choice of modulation scheme and receiver filter can result in better spectrum utilization.

The performance of a first-order digital phase-locked loop (DPLL) using nonuniform sampling is studied in the mobile radio environments. The mobile radio channel is characterized by introducing fast Rayleigh fading and random phase variation to the signal envelope and phase, respectively. The nonlinear stochastic difference equation describing the loop operation in fading environments is introduced. The joint probability density function (pdf) of the random variables of this equation is derived by transformation of random variables for fast Rayleigh fading channels. A closed-form expression for the transition probability of the ChapmanKolmogorov (C-K) equation is obtained for phase step plus noise input and for frequency step plus noise input. The probability density function of the steady-state phase error is obtained by solving the C-K equation numerically.

New techniques for acquiring the codes used in frequency hopping (FH), hybrid frequency hopping/time hopping (FH/TH), and frequency hopping/direct sequence (FH/DS) spread-spectrum systems are presented. Autoregressive spectral estimation is employed to recover the FH code, quadrature processing is used for the DS code, while the TH gating code is recovered by a simple threshold test used in conjunction with the adaptive filter used for the spectral estimation. A wide margin of spectral estimation errors can be tolerated since an acquisition logic is used to estimate the FH code phase. The proposed techniques require acquisition times only of the order of the shift-register generator lengths, whereas other schemes of serial and parallel search require acquisition times of the order of the code lengths. Detailed simulation results will identify the working carrier-to-noise ratio (CNR) range for the different techniques presented as well as the various thresholds involved.

An adaptive digital filtering scheme is developed to deal with the problem of improving the signal-to-noise ratio of a signal corrupted by noise when only general {\em a priori} assumptions regarding the signal and the noise are possible. Specifically, the noise is assumed to be white zero-mean and uncorrelated; while the signal is considered to be band-limited, possibly with slowly varying spectrum. The proposed adaptive digital filtering scheme is based upon a class of variable wave digital filters. Adaptation of the digital filter multipliers is accomplished through the use of an identification procedure based on an adaptive spectral estimation method.

This paper presents an algorithmic method for measuring the instantaneous frequency of a uniformly sampled FM signal. The measured parameter, termed digital instantaneous frequency, is defined in a manner similar to that used to describe frequency-modulated, continuous-time signals. The measurements are derived from an adaptive linear prediction spectral estimates. The proposed algorithm is utilized in the development of a digital processor for FM demodulation which operates on a uniformly sampled FM signal, and its output is a sampled sequence of the estimated demodulated message. The performance of the digital processor is demonstrated and compared with that of a conventional FM discriminator.

The multipath channel for communication between an aerospace vehicle and a ground terminal is modeled by a multiplicative first-order Markov process. The multiplicative process is treated as a component of the message model and the discrete-time demodulation algorithms using the extended Kalman nonlinear estimation technique are developed for continuous-time angle-modulated signals. The equivalent baseband form of the demodulator structure is derived. Two examples of the message process are discussed for an FM system. The simulation results are presented for various values of the bandwidth expansion ratio and the additive SNR. The performance of the baseband algorithms is discussed.

A discrete phase state bit-synchronization loop is proposed for synchronization and detection of a binary nonreturn-to-zero (NRZ) process. The general loop model is developed for 11 and 17 phase states with optional sequential loop filter. Loop performance is considered in terms of the phase error density function, rms phase error, and mean time to bit slippage. It is shown that satisfactory loop performance can be achieved at -3 dB SNR through use of sequential loop filters.

The multipath communication channel for space communication is modeled as a multiplicative channel. This paper discusses the effects of the multiplicative channel processes on the symbol error rate for quadrature modulation (QM) digital modulation schemes. An expression for the upper bound on the probability of error is derived and numerically evaluated. The results are compared with those obtained for additive channels.

Quantization and sampling effects on the digital phasedlocked loop (DPLL) structures obtained for demodulation of anglemodulated signals using extended Kalman filter algorithms are investigated for the high signal-to-noise ratio (SNR) case in this paper. First, the problem of quantization is considered. The validity of the uniform white sequence model for quantizer error in the DPLL is established independent of the sampling rate. Simulation results are presented for several quantizer word lengths. Also, an effective SNR is defined which allows prediction of quantized performance from unquantized results. Secondly, minimum sampling requirements for the DPLL are considered. The effect of sampling rate variation upon the predicted phase error covariance is examined. Again, simulation results are presented and compared to the predicted phase error covariance values. This results in an analytical method for determining minimum sampling rates for the DPLL. Minimum sampling rates for quantized DPLL have also been determined using the effective SNR previously defined.

A discrete receiver structure is introduced that consolidates the problems of bit detection and bit synchronization to provide optimal estimates of bit values without explicit knowledge of the transition epoch. Equations are developed for the propagation of the discrete a posteriori density function on an interval-by-interval basis. The general model is developed for a signal corrupted by additive and multiplicative noise, and the receiver structure is shown to be equivalent to a synchronous discrete receiver. The model is applied to on-off-keyed (OOK) modulation in the presence of multipath.

This paper presents the derivation of the recursive algorithms necessary for real-time digital detection ofM-ary known signals that are subject to independent multiplicative and additive Gaussian noises. The motivating application is minimum probability of error detection of digital data-link messages aboard civil aircraft in the earth reflection multipath environment. For each known signal, the detector contains one Kalman filter and one probability computer. The filters estimate the multipath disturbance. The estimates and the received signal drive the probability computers. Outputs of all the computers are compared in amplitude to give the signal decision. The practicality and usefulness of the detector are extensively discussed.

One of the fundamental technical problems in aeronautical digital communications is that of multipath propagation between aircraft and ground terminal. This paper examines in detail a model of the received multipath signal that is useful for application of modern detection and estimation theories. The model treats arbitrary modulation and covers the selective and nonselective cases. The necessarily nonstationary statistics of the received signal are determined from the link geometry and the surface roughness parameters via a Kirchhoff solution. Use of the model in solving detection and estimation problems is outlined. Results of a flight experiment are presented and compared to a digital simulation for verification of the model.

Quasi-optimum digital phase-locked loops (DPLL) are derived utilizing nonlinear estimation theory. Nonlinear approximations are employed to yield realizable loop structures. Baseband equivalent loop gains are derived, which, under high signal-to-noise ratio conditions may be calculated off line. Additional simplifications are made that permit the application of the Kalman filter algorithms to determine the minimum mean-square error (MSE) loop filter. Consideration is given to sampling rate and implementation requirements. Performance is evaluated by a theoretical analysis and by simulation. Theoretical and simulated results are discussed and a comparison to analog results is made.

An approach to the analysis of performance of quasioptimum digital phase-locked loops (DPLL's) is presented. An expression for the characteristic function of the prior error in the state estimate is derived, and from this expression an infinite dimensional equation for the prior error variance is obtained. The prior errorvariance equation is a function of the communication system model and the DPLL gain and is independent of the method used to derive the DPLL gain. Two approximations are discussed for reducing the prior error-variance equation to finite dimension. The effectiveness of one approximation in analyzing DPLL performance is studied.

A nonstationary signal source can be decomposed into subsources if it exhibits certain characteristics. Adaptive predictive coding is considered as a method of extracting these subsources from the composite source in such a way that the overall communication problem can then be viewed as two different, but connected, communication problems: 1) transmission of predictor parameters, and 2) transmission of the subsource using difference signals. An intermediate fidelity criterion is defined that describes the effects of predictor parameter distortion on the reconstruction of the difference signal. Then a rate-distortion bound on the channel requirements for the transmission of the predictor parameters is found, subject to a dual fidelity criterion. The approach is applicable to a wide range of nonstationary signal classes. The speech process is chosen here because of the wide interests in the information content of speech. The signal class consists specifically of selected voiced phrases.

Bounds on the capacity of a class of digital communication channels are derived. Equating the bounds on capacity to rate-distortion functions of (typical) sources in turn produces bounds on the performance of a class of digital communication systems. For ratios of squared quantization level to noise variance much !ess than one, the power requirements for this class of digital communication systems are shown to be within approximately 3 dB of the theoretical optimum.

In the search for new methods to realize stochastic automata, additive Bernoulli noise linear sequential circuits (ABNLSC) are constructed from linear sequential circuits fed with Bernoulli noise. The characteristics of the subclass of stochastic automata realizable by ABNLSC's are found. These characteristics include: commutativity with respect to input sequences; state minimality; doubly stochastic state transition matrices; and the conditions for the existence of asymptotic distribution of the state.

In this paper, a first-order discrete phase-locked loop that tracks the phase of an incoming analog signal at discrete instants of time occurring once per cycle of the carrier is developed. The functional loop components and the operation of the loop are described in detail. The loop model is developed for both the deterministic and noisy case by a nonlinear coupled difference equation with perturbed (or deviating) arguments. The tracking performance for the inputs of the phase step, the phase ramp, and the sinusoidally modulated carrier is determined and the locking conditions for these input transients are derived. Conceptual designs using this first-order digital loop have been given for the demodulation of phase- and frequency-modulated signals.

This paper presents an approach to the optimum digital demodulation of a continuous-time FM signal using stochastic estimation theory. The primary result is a digital phase-locked loop realization possessing performance characteristics that approach those of the analog counterpart. Some practical considerations are presented and simulation results for a first-order message model are presented.

This paper considers the joint detection and estimation of Gauss-Markov processes in white Gaussian noise, where the operations of detection and estimation are strongly coupled. Explicit Bayes' optimal recursive estimation and detection rules are derived, and it is shown that the resulting optimal receiver is amenable to a causal estimator-correlator-type interpretation.

Nonlinear stochastic estimation theory is applied to the problem of obtaining discrete-time demodulator structures for the common types of analog communication signals. A stochastic communication model is presented that can be treated with current results in estimation theory. Approximate algorithms are developed for the estimation problem that yield mathematically realizable receiver structures for the cases of AM, PM, and FM. The digital receivers yield performance characteristics that compare favorably with those of their analog counterparts.

Using the convex combination of random binary variables, stochastic vectors and stochastic matrices are systematically realized. The techniques are extended to the realization of stochastic automata as randomly fed deterministic automata.

Two different problems of estimating a discrete stochastic process, in the face of Markov dependent uncertainty regarding the presence of the process at each stage of the observation sequence, are considered. Recursive Bayes optimal estimator algorithms are derived for the two cases considered, and the differences between them brought out explicitly.

The problem of estimating the amplitude of a sinusoid from noisy random samples is examined. The maximum-likelihood procedure, applied in the strong-signal case, yields a simple (albeit nonlinear) estimate. The mean-squared estimation error is obtained as a function of the sample size as well as the signal-to-noise ratio.

This paper focuses attention upon the design of a processor and memory system which is structured to achieve a satisfactory balance of processor speed and memory speed when both the processor and input–output controller are simultaneously competing for memory service. A mathematical model is developed to investigate the degree to which the processor is capable of overlapping memory references with instruction execution as a function of respective cycle times, the number of instruction "look-aheads," the number of independent memory modules, and input–output traffic. Utilizing this model, design trade-offs and performance indices are graphically examined for a hypothetical system.

This short note focuses attention upon techniques for dynamic memory allocation in multiprogrammed systems which employ the addressing mechanisms of paging and segmentation. Here, event indicators and mathematical tools are presented which supply characterizations of the paging and segmentation addressing processes. It is shown that these statistical characterizations form data bases which can be used to derive Bayesian storage allocation algorithms conditionally based upon usage, demand, and processing history. It is argued that these characterizations, and algorithms similar to those constructed here, provide a flexible basis for efficient memory management in multiprogrammed, and by extension, time-shared environments. Although emphasis is directed to managing primary (main) memory residence, the techniques could be extended to govern memory management for a hierarchy of storage devices.