7.4.

Equalizers in a Communications Receiver

The previous section demonstrated how a generic adaptive equalizer works during training and defined common notation for algorithm design and analysis. This section describes how the equalizer fits into the wireless communications link. Figure 7.1 shows that the received signal includes channel noise. Because the noise nb(t) is present, an equalizer is unable to achieve perfect performance. Thus there is always some residual ISI and some small tracking error. Noise makes Equation (7.4) hard to realize in practice. Therefore, the instantaneous combined frequency response will not always be flat, resulting in some finite prediction error. The prediction error of the equalizer is defined in Equation (7.19).

7.5. Survey of Equalization Techniques

Equalization techniques can be subdivided into two general categories linear and nonlinear equalization. These categories are determined from how the output of an adaptive equalizer is used for subsequent control (feedback) of the equalizer. In general, the analog signal is processed by the decision making device in the receiver. The decision maker determines the value of the digital data bit being received and applies a slicing or thresholding operation (a nonlinear operation) in order to determine the value of d(t) (see Figure 7.1). If d(t) is not used in the feedback path to adapt the equalizer, the equalization is linear. On the other hand, if d(t) is fed back to change the subsequent outputs of the equalizer, the equalization is nonlinear. Many filter structures are used to implement linear and nonlinear equalizers. Further, for each structure, there are numerous algorithms used to adapt the equalizer. Figure 7.3 provides a general categorization of the equalization techniques according to the types, structures, and algorithms used.

Figure 7.3. Classification of equalizers.

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7.6. Linear Equalizers

As mentioned in Section 7.5, a linear equalizer can be implemented as an FIR filter, otherwise known as the transversal filter. This type of equalizer is the simplest type available. In such an equalizer, the current and past values of the received signal are linearly weighted by the filter coefficient and summed to produce the output, as shown in Figure 7.6. If the delays and the tap gains are analog, the continuous output of the equalizer is sampled at the symbol rate and the samples are applied to the decision device. The implementation is, however, usually carried out in the digital domain where the samples of the received signal are stored in a shift register. The output of this transversal filter before a decision is made (threshold detection) is [Kor85] Equation 7.20

7.7. Nonlinear Equalization

Nonlinear equalizers are used in applications where the channel distortion is too severe for a linear equalizer to handle, and are commonplace in practical wireless systems. Linear equalizers do not perform well on channels which have deep spectral nulls in the passband. In an attempt to compensate for the distortion, the linear equalizer places too much gain in the vicinity of the spectral null, thereby enhancing the noise present in those frequencies. Three very effective nonlinear methods have been developed which offer improvements over linear equalization techniques and are used in most 2G and 3G systems. These are [Pro91]:

7.8. Algorithms for Adaptive Equalization

Since an adaptive equalizer compensates for an unknown and time-varying channel, it requires a specific algorithm to update the equalizer coefficients and track the channel variations. A wide range of algorithms exist to adapt the filter coefficients. The development of adaptive algorithms is a complex undertaking, and it is beyond the scope of this text to delve into great detail on how this is done. Excellent references exist which treat algorithm development [Wid85], [Hay86], [Pro91]. This section describes some practical issues regarding equalizer algorithm design, and outlines three of the basic algorithms for adaptive equalization. Though the algorithms detailed in this section are derived for the linear, transversal equalizer, they can be extended to other equalizer structures, including nonlinear equalizers. The performance of an algorithm is determined by various factors which include:

7.9. Fractionally Spaced Equalizers

The equalizers discussed so far have tap spacings at the symbol rate. It is well known that the optimum receiver for a communication signal corrupted by Gaussian noise consists of a matched filter sampled periodically at the symbol rate of the message. In the presence of channel distortion, the matched filter prior to the equalizer must be matched to the channel and the corrupted signal. In practice, the channel response is unknown, and hence the optimum matched filter must be adaptively estimated. A suboptimal solution in which the matched filter is matched to the transmitted signal pulse may result in a significant

degradation in performance. In addition, such a suboptimal filter is extremely sensitive to any timing error in the sampling of its output [Qur77]. A fractionally spaced equalizer (FSE) is based on sampling the incoming signal at least as fast as the Nyquist rate [Pro91]. The FSE compensates for the channel distortion before aliasing effects occur due to the symbol rate sampling. In addition, the equalizer can compensate for any timing delay for any arbitrary timing phase. In effect, the FSE incorporates the functions of a matched filter and equalizer into a single filter structure. Simulation results demonstrating the effectiveness of the FSE over a symbol rate equalizer have been given in the papers by Qureshi and Forney [Qur77], and Gitlin and Weinstein [Git81]. Nonlinear equalizers based on MLSE techniques appear to be gaining popularity in modern wireless systems (these were described in Section 7.7.2). The interested reader may find Chapter 6 of [Ste94] useful for further work in this area.

7.10. Diversity Techniques

Diversity is a powerful communication receiver technique that provides wireless link improvement at relatively low cost. Unlike equalization, diversity requires no training overhead since a training sequence is not required by the transmitter. Furthermore, there are a wide range of diversity implementations, many which are very practical and provide significant link improvement with little added cost. Diversity exploits the random nature of radio propagation by finding independent (or at least highly uncorrelated) signal paths for communication. In virtually all applications, diversity decisions are made by the receiver, and are unknown to the transmitter.

7.11. RAKE Receiver

In CDMA spread spectrum systems (see Chapter 6), the chip rate is typically much greater than the flat-fading bandwidth of the channel. Whereas conventional modulation techniques require an equalizer to undo the intersymbol interference between adjacent symbols, CDMA spreading codes are designed to provide very low correlation between successive chips. Thus, propagation delay spread in the radio channel merely provides multiple versions of the transmitted signal at the receiver. If these multipath components are delayed in time by more than a chip duration, they appear like uncorrelated noise at a CDMA receiver, and equalization is not required. The spread spectrum processing gain makes uncorrelated noise negligible after despreading.

However, since there is useful information in the multipath components, CDMA receivers may combine the time delayed versions of the original signal transmission in order to improve the signal-to-noise ratio at the receiver. A RAKE receiver does just this it attempts to collect the time-shifted versions of the original signal by providing a separate correlation receiver for each of the multipath signals. Each correlation receiver may be adjusted in time delay, so that a microprocessor controller can cause different correlation receivers to search in different time windows for significant multipath. The range of time delays that a particular correlator can search is called a search window. The RAKE receiver, shown in Figure 7.16, is essentially a diversity receiver designed specifically for CDMA, where the diversity is provided by the fact that the multipath components are practically uncorrelated from one another when their relative propagation delays exceed a chip period.

7.12. Interleaving
Interleaving is used to obtain time diversity in a digital communications system without adding any overhead. Interleaving has become an extremely useful technique in all second and third generation wireless systems, due to the rapid proliferation of digital speech coders which transform analog voices into efficient digital messages that are transmitted over wireless links (speech coders are presented in Chapter 8). Because speech coders attempt to represent a wide range of voices in a uniform and efficient digital format, the encoded data bits (called source bits) carry a great deal of information, and as explained in Chapters 8 and 11, some source bits are more important than others and must be protected from errors. It is typical for many speech coders to produce several important bits in succession, and it is the function of the interleaver to spread these bits out in time so that if there is a deep fade or noise burst, the important bits from a block of source data are not corrupted at the same time. By spreading the source bits over time, it becomes possible to make use of error control coding (calledchannel coding) which protects the source data from corruption by the channel. Since error control codes are designed to protect against channel errors that may occur randomly or in a bursty manner, interleavers scramble the time order of source bits before they are channel coded.