Intersymbol Interference Cancellation Research Articles

Analysis of predistortion, equalization, and ISI cancellation techniques in digital radio systems with nonlinear transmit amplifiers

An analysis made of the performance of predistortion, equalization, and intersymbol interference (ISI) cancellation techniques in compensating for the transmit amplifier nonlinearity in digital microwave radio systems. The study is carried out using the 64 QAM and 256 QAM signal formats and two values of the roll-off factor in the Nyquist pulse shaping. The simulated compensation techniques include three types of predistortion, two ISI cancellers, and several nonlinear equalizers with or without decision feedback. A basic result is that decision-feedback equalizers do not offer any significant advantage over nonrecursive equalizers. It is also shown that ISI cancelers with a memoryless equalizer as first-stage decision device do not perform any better than nonlinear equalizers of similar complexity. Another contribution an improved fifth-order analog signal predistortion technique is analyzed. The gain that can be achieved using a modified 256 QAM signal constellation that is more robust to nonlinear distortion is quantified. >

Digital Subscriber Line Terminals for Use with Correlated Line Codes

This paper is a mathematical analysis of digital subscriber lines that use adaptive decision feedback equalization (DFE). Adaptive transversal feedforward filters (FFF) and feedback filters (FBF) with baud rate sampling are assumed. In order that all block codes are considered in the model, correlation between the values of the transmitted symbol values is included. Additionally, added noise is not assumed to be white, to permit the eyaluation of the effects of near end crosstalk and residual echo noise. It is shown that the correlation between the transmitted symbols causes the FBF to adapt differently than they usually do (i.e., cancel intersymbol interference within the range of the FBF) in two ways: the FBF uses the correlation between transmitted symbols to reduce the intersymbol interference outside of its range, and a bias vector is added to the FBF tap weights that adjusts future slicing levels to account for the variable likelihood of future transmitted symbol values. It is found that the received signal-to-noise ratio is approximately a linear function of loop insertion loss when the loss is high and noise is low.

Combination of cancelling intersymbol interference and decoding of error-correcting code

The paper proposes and investigates a new type of nonlinear adaptive canceller of intersymbol interference (ISI) called ‘ACDEC’. The ACDEC uses an automatic equaliser including a decoder of an error-correcting code (AEDEC), to obtain more reliable preliminary decisions than those by the linear equaliser used in the linear adaptive canceller of Gersho and Lim. By using the modified structure and incorporating one more decoder, the tap values of both the canceller's delay line and the adaptive matched filter are correctively updated with reduced final decision errors in the same manner as AEDEC. The theoretical calculation of the error probability in the joint repetitive scheme of both cancelling ISI and decoding indicates that two repetitions, i.e. ACDEC, can provide considerable improvement in the error probability performance in the steady state without excessive complexity of implementation. Moreover, computer simulations show that an ACDEC achieves stable convergence and exhibits better adaptability in noisy and time-varient channels.

Adaptive equalization

Bandwidth-efficient data transmission over telephone and radio channels is made possible by the use of adaptive equalization to compensate for the time dispersion introduced by the channel Spurred by practical applications, a steady research effort over the last two decades has produced a rich body of literature in adaptive equalization and the related more general fields of reception of digital signals, adaptive filtering, and system identification. This tutorial paper gives an overview of the current state of the art in adaptive equalization. In the first part of the paper, the problem of intersymbol interference (ISI) and the basic concept of transversal equalizers are introduced followed by a simplified description of some practical adaptive equalizer structures and their properties. Related applications of adaptive filters and implementation approaches are discussed. Linear and nonlinear receiver structures, their steady-state performance and sensitivity to timing phase are presented in some depth in the next part. It is shown that a fractionally spaced equalizer can serve as the optimum receive filter for any receiver. Decision-feedback equalization, decision-aided ISI cancellation, and adaptive filtering for maximum-likelihood sequence estimation are presented in a common framework. The next two parts of the paper are devoted to a discussion of the convergence and steady-state properties of least mean-square (LMS) adaptation algorithms, including digital precision considerations, and three classes of rapidly converging adaptive equalization algorithms: namely, orthogonalized LMS, periodic or cyclic, and recursive least squares algorithms. An attempt is made throughout the paper to describe important principles and results in a heuristic manner, without formal proofs, using simple mathematical notation where possible.

Adaptive Cancellation of Nonlinear Intersymbol Interference for Voiceband Data Transmission

Two related cancellation techniques based on tentative decisions are proposed for removing intersymbol interference, due to channel nonlinearities, for voiceband data transmission. Cancellation of intersymbol interference avoids the need for equalizing or inverting analog nonlinear operations as in the decision-feedback method previously proposed by Falconer. The nonlinear cancellation approach removes nonlinear interference terms without excessive noise enhancement and allows effective implementation with ROM's to generate the needed nonlinear signal variables that are to be weighted and summed. The first technique can be incorporated as an add-on unit to a present-day modem; the second technique requires a different receiver architecture but offers somewhat enhanced improvement. Simulation results for 9.6 kbit/s data transmission over nonlinear channel models show that both techniques can offer substantial performance improvements for 16-point QAM systems.

Adaptive equalization

Bandwidth-efficient data transmission over telephone and radio channels is made possible by the use of adaptive equalization to compensate for the time dispersion introduced by the channel Spurred by practical applications, a steady research effort over the last two decades has produced a rich body of literature in adaptive equalization and the related more general fields of reception of digital signals, adaptive filtering, and system identification. This tutorial paper gives an overview of the current state of the art in adaptive equalization. In the first part of the paper, the problem of intersymbol interference (ISI) and the basic concept of transversal equalizers are introduced followed by a simplified description of some practical adaptive equalizer structures and their properties. Related applications of adaptive filters and implementation approaches are discussed. Linear and nonlinear receiver structures, their steady-state performance and sensitivity to timing phase are presented in some depth in the next part. It is shown that a fractionally spaced equalizer can serve as the optimum receive filter for any receiver. Decision-feedback equalization, decision-aided ISI cancellation, and adaptive filtering for maximum-likelihood sequence estimation are presented in a common framework. The next two parts of the paper are devoted to a discussion of the convergence and steady-state properties of least mean-square (LMS) adaptation algorithms, including digital precision considerations, and three classes of rapidly converging adaptive equalization algorithms: namely, orthogonalized LMS, periodic or cyclic, and recursive least squares algorithms. An attempt is made throughout the paper to describe important principles and results in a heuristic manner, without formal proofs, using simple mathematical notation where possible.

Adaptive Cancellation of Intersymbol Interference for Data Transmission

In this paper, we analyze a technique for accurately detecting transmitted data symbols contained in a modulated signal that has been degraded by a linear dispersive channel and additive Gaussian noise. The approach uses an adaptive equalizer which provides preliminary decisions to an adaptive canceller. The canceller output is used to remove the interference from an adaptive matching filter, resulting in the desired signal. Channel equalization attempts to invert the channel transfer function, while avoiding excessive noise enhancement. However, cancellation (as used in echo cancellers), attempts to generate a replica of the interfering signal and subtract it from the actual received signal containing the sum of the desired signal and interference. The cancellation approach, unlike equalization, offers the possibility of removing interference without enhancing the level of noise already present in the received waveform. Simulation results for transmission over practical channels show significant improvement of linear cancellation over both linear forward and decision-feedback equalization.

An intersymbol interference cancellation equaliser for use in systems employing envelope detection

A new scheme for the adaptive equalisation of incoherently demodulated signals is presented. In the scheme the output from an envelope detector is fed into a tapped-delay-line filter and simultaneously into a simple phase-estimating circuit. An estimate of the intersymbol interference is obtained and this is then subtracted from an estimate of the signal appearing at the input to the envelope detector to obtain an estimate of the transmitted data. The theoretical basis for the operation of the system is derived, and some results obtained when using the system to equalise time-varying channels are presented.

An Equalizer Structure with Reduced Sampling Time Reference Sensitivity

In the last decade the work on linear tapped delay line equalizers has proved successful in reducing intersymbol interferenceone of the largest problems in data communication systems. An adjoining problem is the selection of a sampling time reference. An equalizer structure less sensitive to sampling time reference is studied for adaptive equalization. The equalization is simply done by means of tap coefficient adjustment. The equalizer consists of two parallel branches, each containing a transversal filter with adjustable tap coefficients. The branches are connected by a fixed filter, the transfer function of which is selected such that the equalizer can perform a sampling time reference displacement and a cancellation of intersymbol interference simultaneously. The criterion of goodness is the conventional meansquare error (including noise) between the actual output and a desired output. The equalizer is applied in two examples showing-in contrast to a conventional equalizer with the same total number of tap coefficients-the insensitivity of the minimum mean-square error (MMSE) to how the the sampling time reference is selected. Thus no extra circuits are needed for this purpose, while a mean-square error that is as low as the best that can be obtained with a conventional equalizer is adaptively maintained.