The general technique of parallel transmission on many carriers, called multicarrier modulation (MCM), is explained. The performance that can be achieved on an undistorted channel and algorithms for achieving that performance are discussed. Ways of dealing with channel impairments and of improving the performance through coding are described, and implementation methods are considered. Duplex operation of MCM and the possible use of this on the general switched telephone network are examined. >

Multicarrier modulation for data transmission: an idea whose time has come

This paper discusses the analysis and simulation of a technique for combating the effects of multipath propagation and cochannel interference on a narrow-band digital mobile channel. This system uses the discrete Fourier transform to orthogonally frequency multiplex many narrow subchannels, each signaling at a very low rate, into one high-rate channel. When this technique is used with pilot-based correction, the effects of flat Rayleigh fading can be reduced significantly. An improvement in signal-to-interference ratio of 6 dB can be obtained over the bursty Rayleigh channel. In addition, with each subchannel signaling at a low rate, this technique can provide added protection against delay spread. To enhance the behavior of the technique in a heavily frequency-selective environment, interpolated pilots are used. A frequency offset reference scheme is employed for the pilots to improve protection against cochannel interference.

Analysis and Simulation of a Digital Mobile Channel Using Orthogonal Frequency Division Multiplexing

This paper discusses the effects of frequency offset on the performance of orthogonal frequency division multiplexing (OFDM) digital communications. The main problem with frequency offset is that it introduces interference among the multiplicity of carriers in the OFDM signal. It is shown, and confirmed by simulation, that to maintain signal-to-interference ratios of 20 dB or greater for the OFDM carriers, offset is limited to 4% or less of the intercarrier spacing. Next, the paper describes a technique to estimate frequency offset using a repeated data symbol. A maximum likelihood estimation (MLE) algorithm is derived and its performance computed and compared with simulation results. Since the intercarrier interference energy and signal energy both contribute coherently to the estimate, the algorithm generates extremely accurate estimates even when the offset is far too great to demodulate the data values. Also, the estimation error depends only on total symbol energy so it is insensitive to channel spreading and frequency selective fading. A strategy is described for initial acquisition in the event of uncertainty in the initial offset that exceeds 1/2 the carrier spacing, the limit of the MLE algorithm. >

A technique for orthogonal frequency division multiplexing frequency offset correction

High peak-to-average power ratio of the transmit signal is a major drawback of multicarrier transmission such as OFDM or DMT. This article describes some of the important PAPR reduction techniques for multicarrier transmission including amplitude clipping and filtering, coding, partial transmit sequence, selected mapping, interleaving, tone reservation, tone injection, and active constellation extension. Also, we make some remarks on the criteria for PAPR reduction technique selection and briefly address the problem of PAPR reduction in OFDMA and MIMO-OFDM.

An overview of peak-to-average power ratio reduction techniques for multicarrier transmission

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.

https://ieeexplore.ieee.org/iel5/35/23767/01090994.pdf

Adaptive equalization

This paper presents a principle of orthogonal multiplexing for transmitting a number of data messages simultaneously through a linear band-limited transmission medium at a maximum data rate without interchannel and intersymbol interferences. A general method is given for synthesizing an infinite number of classes of band-limited orthogonal time functions in a limited frequency band. Stated in practical terms, the method permits the synthesis of a large class of practical transmitting filter characteristics for an arbitrarily given amplitude characteristic of the transmission medium. Rectangular-shaped ideal filters are not required. The synthesis procedure is convenient. Furthermore, the amplitude and the phase characteristics of the transmitting filters can be synthesized independently. Adaptive correlation reception can be used for data processing, since the received signals remain orthogonal no matter what the phase distortion is in the transmission medium. The system provides the same signal distance protection against channel noises as if the signals of each channel were transmitted through an independent medium and intersymbol interference in each channel were eliminated by reducing data rate.

Synthesis of band-limited orthogonal signals for multichannel data transmission

The use of transversal filters for automatic equalization has made possible high-speed data communication over voiceband telephone channels. Recently much attention has been focused on the possible use of the high-speed data sets in private line multiparty polling systems. However, for such applications, it is necessary to reduce the start-up time of the present automatic equalizer drastically. This paper examines the start-up time (settling time) of the transversal filter equalizer for two important classes of data communication systems: Class IV partial-response systems and single-sideband Nyquist systems. (The latter represents the limiting case of vestigial-sideband systems with small roll-off bandwidth.) It is shown that in single-sideband Nyquist systems the input signals to the gain controls of the transversal equalizer may be nearly orthonormal. Consequently the equalizers may have a short settling time. It is also shown that the equalizer settling time is much longer in Class IV partial-response systems, because such systems use controlled intersymbol interference and the input signals to the gain controls are highly correlated. The possibility of reducing the settling time of the automatic equalizers is examined. A new equalizer structure is developed based on the following principles: (i) Equalizer settling time can be minimized by making the input signals to the gain controls orthonormal, and (ii) Such a minimization does not change the noise power, the mean-square equalization error, the convexity of the gain control adjustment, and the feedback control loops in the equalizer. These principles are general in that they apply regardless of the type of modulation — single-, vestigial-, or double-sideband (SSB, VSB, or DSB) — or the signaling scheme (Nyquist or partial-response). Application of these principles to Class IV partial-response systems is considered. For private line systems and systems where amplitude distortions in the communication channels are not severe (delay distortions can be arbitrary), the new equalizer can be implemented by simply adding a prefixed weighting matrix to the conventional transversal equalizer. Analysis and computer simulation show that the use of such a new equalizer can residt in a significant reduction in the system's start-up time.

A new equalizer structure for fast start-up digital communication

In this paper, we analyze single-sideband amplitude-modulation digital communication systems to develop a method for jointly and optimally setting the carrier phase and the automatic transversal equalizer of such systems. Mean-square equalization error is used as the performance criterion. We develop a simple receiver structure and study the convergence of the method. Exact locations of the stationary points in the parameter space are determined and the classifications of the stationary points are obtained. We show that the mean-square equalization error has only global minima and saddlepoints, but not local minima and maxima. Thus, the mean-square equalization error will converge to the absolute minimum by the proposed method, regardless of the initial settings of the parameters. A simple condition on the step sizes of the adjustments is also obtained which ensures the convergence of the process. Explicit formulas of the joint optimum parameter settings and of the corresponding minimum mean-square error are obtained. For illustration purposes, a single-sideband digital communication system using a five- or nine-tap transversal equalizer is simulated on a computer. Both theory and simulation show that the equalization error depends critically on the carrier phase when the number of equalizer taps is not large, and that the minimum equalization error can be obtained by using the proposed method.

Joint optimization of automatic equalization and carrier acquisition for digital communication

The information rate of a hybrid coaxial cable transmission system using multilevel pulse amplitude modulation is studied, assuming that the additive repeater noise has a flat spectral density and that statistically independent message symbols are transmitted. Questions considered theoretically are: (i) Reduction in information rate when some repeaters in an “all digital repeater” system are replaced by analog repeaters, (ii) Number of digital repeaters required for converting an analog system to digital service, (iii) Information rate versus number of added analog repeaters in a fixed digital repeater section, (iv) System sensitivity to repeater output power and noise spectral density variations, and (v) Bit rate versus baud rate and achieving the greatest bit rate. Curves and tables answer these questions. It is economical and theoretically optimum to use identical analog repeaters and uniform repeater spacing for the coaxial cable systems considered. The optimum gain-frequency characteristic for the analog repeaters is the same for both analog and digital transmission. Analog cable systems can be adapted directly to hybrid digital service with no compromise in theoretical performance.

http://ieeexplore.ieee.org/iel7/6731005/6774030/06774037.pdf

Hybrid digital transmission systems part II: Information rate of hybrid coaxial cable systems

A hybrid digital transmission system consists of analog repeaters placed between digital repeaters. Joint optimization of the analog and digital repeaters is considered in this paper, using minimum mean-square error between the transmitted and received symbols as the performance criterion. A general hybrid system is considered. The joint optimization problem is solved in closed form for deterministic sampling under two usually satisfied conditions. From the results the minimum mean-square error and the optimum repeater characteristics can be computed for given system parameters. Timing error is also considered. From a general result, it is concluded that in many practical systems it is not only economical, but also optimum, to use identical analog repeaters, and that hybrid systems can be used for either digital or voice transmission with no compromise in theoretical performance.

http://ieeexplore.ieee.org/iel7/6731005/6774030/06774036.pdf

Robert W. Chang

Papers

Synthesis of band-limited orthogonal signals for multichannel data transmission

A new equalizer structure for fast start-up digital communication

Joint optimization of automatic equalization and carrier acquisition for digital communication

Hybrid digital transmission systems part II: Information rate of hybrid coaxial cable systems

Hybrid digital transmission systems part I: Joint optimization of analog and digital repeaters