This paper is motivated by the need for fundamental understanding of ultimate limits of bandwidth efficient delivery of higher bit-rates in digital wireless communications and to also begin to look into how these limits might be approached. We examine exploitation of multi-element array (MEA) technology, that is processing the spatial dimension (not just the time dimension) to improve wireless capacities in certain applications. Specifically, we present some basic information theory results that promise great advantages of using MEAs in wireless LANs and building to building wireless communication links. We explore the important case when the channel characteristic is not available at the transmitter but the receiver knows (tracks) the characteristic which is subject to Rayleigh fading. Fixing the overall transmitted power, we express the capacity offered by MEA technology and we see how the capacity scales with increasing SNR for a large but practical number, n, of antenna elements at both transmitter and receiver.

We investigate the case of independent Rayleigh faded paths between antenna elements and find that with high probability extraordinary capacity is available. Compared to the baseline n = 1 case, which by Shannon‘s classical formula scales as one more bit/cycle for every 3 dB of signal-to-noise ratio (SNR) increase, remarkably with MEAs, the scaling is almost like n more bits/cycle for each 3 dB increase in SNR. To illustrate how great this capacity is, even for small n, take the cases n = 2, 4 and 16 at an average received SNR of 21 dB. For over 99% of the channels the capacity is about 7, 19 and 88 bits/cycle respectively, while if n = 1 there is only about 1.2 bit/cycle at the 99% level. For say a symbol rate equal to the channel bandwith, since it is the bits/symbol/dimension that is relevant for signal constellations, these higher capacities are not unreasonable. The 19 bits/cycle for n = 4 amounts to 4.75 bits/symbol/dimension while 88 bits/cycle for n = 16 amounts to 5.5 bits/symbol/dimension. Standard approaches such as selection and optimum combining are seen to be deficient when compared to what will ultimately be possible. New codecs need to be invented to realize a hefty portion of the great capacity promised.

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On Limits of Wireless Communications in a Fading Environment when UsingMultiple Antennas

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

The use of infrared radiation as a medium for high-speed short-range wireless digital communication is discussed. Available infrared links and local-area networks are described. Advantages and drawbacks of the infrared medium are compared to those of radio and microwave media. The physical characteristics of infrared channels using intensity modulation with direct detection (IM/DD) are presented including path losses and multipath responses. Natural and artificial ambient infrared noise sources are characterized. Strategies for designs of transmitter and receivers that maximize link signal-to-noise ratio (SNR) are described. Several modification formats are discussed in detail, including on-off keying (OOK) pulse-position modulation (PPM), and subcarrier modulation. The performance of these techniques in the presence of multipath distortion is quantified. Techniques for multiplexing the transmissions of different users are reviewed. The performance of an experimental 50-Mb/s on-off-keyed diffuse infrared link is described.

Wireless Infrared Communications

Conventional equalization and carrier recovery algorithms for minimizing mean-square error in digital communication systems generally require an initial training period during which a known data sequence is transmitted and properly synchronized at the receiver. This paper solves the general problem of adaptive channel equalization without resorting to a known training sequence or to conditions of limited distortion. The criterion for equalizer adaptation is the minimization of a new class of nonconvex cost functions which are shown to characterize intersymbol interference independently of carrier phase and of the data symbol constellation used in the transmission system. Equalizer convergence does not require carrier recovery, so that carrier phase tracking can be carried out at the equalizer output in a decision-directed mode. The convergence properties of the self-recovering algorithms are analyzed mathematically and confirmed by computer simulation.

Self-Recovering Equalization and Carrier Tracking in Two-Dimensional Data Communication Systems

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

A speaker verification and voice command system utilizing speech templates stored in an integrated circuit card is disclosed. To verify the user's identity, a comparison is made between a plurality of reference speech templates stored in the user's integrated circuit card and a test template formed from a word or words spoken by the user.

Speaker verification system using integrated circuit cards

A fractionally spaced equalizer is a nonrecursive adaptive filter whose tap weights are spaced a fraction of a symbol interval apart. Such an equalizer can significantly enhance modem performance in the presence of severe linear distortion, when compared with a conventional synchronous equalizer whose taps are spaced a symbol interval apart. However, a digitally implemented, fractionally spaced equalizer generally will exhibit long-term instability when the conventional tap-adjustment algorithm is used. This occurs because, in contrast to the synchronous equalizer, a fractionally spaced equalizer generally will have many sets of tap values, which result in nearly equal values of mean-squared error (mse). Some of these tap settings — which invariably will be attained because of biases in the digital tap-updating circuitry — are large enough to cause register overflows and consequent performance deterioration. In this paper we report how a simple modification in the tap-adjustment algorithm provides a solution to the above problem. The modified tap-adjustment algorithm prevents the buildup of large coefficient values by systematically “leaking” or decreasing the magnitudes of all the equalizer tap weights. For an experimental modem operating at 9.6 kb/s, it has been demonstrated that the tap-leakage adjustment algorithm prevents the accumulation of large equalizer tap values, while permitting the full performance gain of a fractionally spaced equalizer to be realized.

The tap-leakage algorithm: An algorithm for the stable operation of a digitally implemented, fractionally spaced adaptive equalizer

A long-standing communications problem is the efficient coding of a block of binary data into a pair of in-phase and quadrature components. This modulation technique may be regarded as the placing of a discrete number of signal points in two dimensions. Quadrature amplitude modulation (QAM) and combined amplitude and phase modulation (AM-PM) are two familiar examples of this signaling format. Subject to a peak or average power constraint, the selection of the signal coordinates is done so as to minimize the probability of error. In the design of high-speed data communication systems this problem becomes one of great practical significance since the dense packing of signal points reduces the margin against Gaussian noise. Phase jitter, which tends to perturb the angular location of the transmitted signal point, further degrades the error rate. Previous investigations have considered the signal evaluation and design problem in the presence of Gaussian noise alone and within the framework of a particular structure, such as conventional amplitude and phase modulation. We present techniques to evaluate and optimize the choice of a signal constellation in the presence of both Gaussian noise and carrier phase jitter. The performance of a number of currently used or proposed signal constellations are compared. The evaluation and the optimization are based upon a perturbation analysis of the probability density of the received signal given the transmitted signal. Laplace's asymptotic formula is used for the evaluation. Discretizing the signal space reduces the optimal signal design problem under a peak power constraint to a tractable mathematical programming problem. Our results indicate that in Gaussian noise alone an improvement in signal-to-noise ratio of as much as 2 dB may be realized by using quadrature amplitude modulation instead of conventional amplitude and phase modulation. New modulation formats are proposed which perform very well in Gaussian noise and additionally are quite insensitive to moderate amounts of phase jitter.

On the selection of a two-dimensional signal constellation in the presence of phase jitter and Gaussian noise

An adaptive echo canceller for two-wire, simultaneous two-way data communication at full bandwidth uses Nyquist-interval, rather than baud-interval, processing to achieve independence from timing discrepancies between near-end and far-end terminals. The entire echo signal, and not merely baud-interval samples thereof, is suppressed. The echo canceller is preferably a transversal structure.

Echo cancellation in two-wire, two-way data transmission systems

A digitally-implemented echo canceller operating at a rate greater than twice the highest passband frequency is proposed for fullduplex data transmission on a two-wire circuit. Located at each end of the communication circuit, the cancellers operate independently of the local receivers and do not require synchronization of the two data stations. Economies in storage and A/D conversion, in comparison with voice-type cancellers described in the literature, are achieved by accepting data symbols as reference input instead of samples of the transmitted line signal. Convergence of transversal filter tap weights is demonstrated under a mean-squared error criterion, and use of the real passband error rather than the complex analytic error is found to lead to the same residual error at the expense of a doubled convergence time. An operational protocol and adaptation algorithm are proposed which make possible both rapid start-up and slower adaptation during double talking. Provision is made for limiting the number and location of active taps on the transversal filter to those actually necessary for replicating the echo channel, resulting in two transversal filter sections of moderate length separated by a bulk delay. Results from a computer simulation of the proposed canceller are offered to demonstrate that convergence of mean-squared tap-weight error follows the predicted exponential characteristic, that the length of the bulk delay can be determined from a single channel sounding under typical channel noise conditions, and that the use of an averaged-gradient algorithm will allow the canceller to adapt, although slowly, to a change in the echo channel during full-duplex operation.

Stephen B. Weinstein

Papers

Speaker verification system using integrated circuit cards

The tap-leakage algorithm: An algorithm for the stable operation of a digitally implemented, fractionally spaced adaptive equalizer

On the selection of a two-dimensional signal constellation in the presence of phase jitter and Gaussian noise

Echo cancellation in two-wire, two-way data transmission systems

A Passband Data-Driven Echo Canceller for Full-Duplex Transmission on Two-Wire Circuits