In Chapter 3, we established a framework for demodulation over AWGN channels under the assumption that the receiver knows and can reproduce the noiseless received signal for each possible transmitted signal. These provide “templates” against which we can compare the noisy received signal (using correlation), and thereby make inferences about the likelihood of each possible transmitted signal. Before the receiver can arrive at these templates, however, it must estimate unknown parameters such as the amplitude, frequency and phase shifts induced by the channel. We discuss synchronization techniques for obtaining such estimates in this chapter. Alternatively, the receiver might fold in implicit estimation of these parameters, or average over the possible values taken by these parameters, in the design of the demodulator. Noncoherent demodulation, discussed in detail in this chapter, is an example of such an approach to dealing with unknown channel phase. Noncoherent communication is employed when carrier synchronization is not available (e.g., because of considerations of implementation cost or complexity, or because the channel induces a difficult-to-track time-varying phase, as for wireless mobile channels). Noncoherent processing is also an important component of many synchronization algorithms (e.g., for timing synchronization, which often takes place prior to carrier synchronization).

Since there are many variations in individual (and typically proprietary) implementations of synchronization and demodulation algorithms, the focus here is on developing basic principles, and on providing some simple examples of how these principles might be applied.

In this chapter, we develop channel equalization techniques for handling the intersymbol interference (ISI) incurred by a linearly modulated signal that goes through a dispersive channel. The principles behind these techniques also apply to dealing with interference from other users, which, depending on the application, may be referred to as co-channel interference, multiple-access interference, multiuser interference, or crosstalk. Indeed, we revisit some of these techniques in Chapter 8 when we briefly discuss multiuser detection. More generally, there is great commonality between receiver techniques for efficiently accounting for memory, whether it is introduced by nature, as considered in this chapter, or by design, as in the channel coding schemes considered in Chapter 7. Thus, the optimum receiver for ISI channels (in which the received signal is a convolution of the transmitted signal with the channel impulse response) uses the same Viterbi algorithm as the optimum receiver for convolutional codes (in which the encoded data are a convolution of the information stream with the code “impulse response”) in Chapter 7.

The techniques developed in this chapter apply to single-carrier systems in which data are sent using linear modulation. An alternative technique for handling dispersive channels, discussed in Chapter 8, is the use of multicarrier modulation, or orthogonal frequency division multiplexing (OFDM). Roughly speaking, OFDM, or multicarrier modulation, transforms a system with memory into a memoryless system in the frequency domain, by decomposing the channel into parallel narrowband subchannels, each of which sees a scalar channel gain.

Information theory (often termed Shannon theory in honor of its founder, Claude Shannon) provides fundamental benchmarks against which a communication system design can be compared. Given a channel model and transmission constraints (e.g., on power), information theory enables us to compute, at least in principle, the highest rate at which reliable communication over the channel is possible. This rate is called the channel capacity.

Once channel capacity is computed for a particular set of system parameters, it is the task of the communication link designer to devise coding and modulation strategies that approach this capacity. After 50 years of effort since Shannon's seminal work, it is now safe to say that this goal has been accomplished for some of the most common channel models. The proofs of the fundamental theorems of information theory indicate that Shannon limits can be achieved by random code constructions using very large block lengths. While this appeared to be computationally infeasible in terms of both encoding and decoding, the invention of turbo codes by Berrou et al. in 1993 provided implementable mechanisms for achieving just this. Turbo codes are random-looking codes obtained from easy-to-encode convolutional codes, which can be decoded efficiently using iterative decoding techniques instead of ML decoding (which is computationally infeasible for such constructions). Since then, a host of “turbo-like” coded modulation strategies have been proposed, including rediscovery of the low density parity check (LDPC) codes invented by Gallager in the 1960s.

We now know that information is conveyed in a digital communication system by selecting one of a set of signals to transmit. The received signal is a distorted and noisy version of the transmitted signal. A fundamental problem in receiver design, therefore, is to decide, based on the received signal, which of the set of possible signals was actually sent. The task of the link designer is to make the probability of error in this decision as small as possible, given the system constraints. Here, we examine the problem of receiver design for a simple channel model, in which the received signal equals one of M possible deterministic signals, plus white Gaussian noise (WGN). This is called the additive white Gaussian noise (AWGN) channel model. An understanding of transceiver design principles for this channel is one of the first steps in learning digital communication theory. White Gaussian noise is an excellent model for thermal noise in receivers, whose PSD is typically flat over most signal bandwidths of interest.

In practice, when a transmitted signal goes through a channel, at the very least, it gets attenuated and delayed, and (if it is a passband signal) undergoes a change of carrier phase. Thus, the model considered here applies to a receiver that can estimate the effects of the channel, and produce a noiseless copy of the received signal corresponding to each possible transmitted signal.

Communication has been one of the deepest needs of the human race throughout recorded history. It is essential to forming social unions, to educating the young, and to expressing a myriad of emotions and needs. Good communication is central to a civilized society.

The various communication disciplines in engineering have the purpose of providing technological aids to human communication. One could view the smoke signals and drum rolls of primitive societies as being technological aids to communication, but communication technology as we view it today became important with telegraphy, then telephony, then video, then computer communication, and today the amazing mixture of all of these in inexpensive, small portable devices.

Initially these technologies were developed as separate networks and were viewed as having little in common. As these networks grew, however, the fact that all parts of a given network had to work together, coupled with the fact that different components were developed at different times using different design methodologies, caused an increased focus on the underlying principles and architectural understanding required for continued system evolution.

This need for basic principles was probably best understood at American Telephone and Telegraph (AT&T), where Bell Laboratories was created as the research and development arm of AT&T. The Math Center at Bell Labs became the predominant center for communication research in the world, and held that position until quite recently.

Introduction

Chapter 7 showed how to characterize noise as a random process. This chapter uses that characterization to retrieve the signal from the noise-corrupted received waveform. As one might guess, this is not possible without occasional errors when the noise is unusually large. The objective is to retrieve the data while minimizing the effect of these errors. This process of retrieving data from a noise-corrupted version is known as detection.

Detection, decision making, hypothesis testing, and decoding are synonyms. The word detection refers to the effort to detect whether some phenomenon is present or not on the basis of observations. For example, a radar system uses observations to detect whether or not a target is present; a quality control system attempts to detect whether a unit is defective; a medical test detects whether a given disease is present. The meaning of detection has been extended in the digital communication field from a yes/no decision to a decision at the receiver between a finite set of possible transmitted signals. Such a decision between a set of possible transmitted signals is also called decoding, but here the possible set is usually regarded as the set of codewords in a code rather than the set of signals in a signal set. Decision making is, again, the process of deciding between a number of mutually exclusive alternatives.

Introduction

A general block diagram of a point-to-point digital communication system was given in Figure 1.1. The source encoder converts the sequence of symbols from the source to a sequence of binary digits, preferably using as few binary digits per symbol as possible. The source decoder performs the inverse operation. Initially, in the spirit of source/channel separation, we ignore the possibility that errors are made in the channel decoder and assume that the source decoder operates on the source encoder output.

We first distinguish between three important classes of sources.

1. • Discrete sources The output of a discrete source is a sequence of symbols from a known discrete alphabet X. This alphabet could be the alphanumeric characters, the characters on a computer keyboard, English letters, Chinese characters, the symbols in sheet music (arranged in some systematic fashion), binary digits, etc. The discrete alphabets in this chapter are assumed to contain a finite set of symbols.

2. It is often convenient to view the sequence of symbols as occurring at some fixed rate in time, but there is no need to bring time into the picture (for example, the source sequence might reside in a computer file and the encoding can be done off-line).

3. This chapter focuses on source coding and decoding for discrete sources. Supplementary references for source coding are given in Gallager (1968, chap. 3) and Cover and Thomas (2006, chap. 5). A more elementary partial treatment is given in Proakis and Salehi (1994, sect. 4.1–4.3).

4. […]

Introduction

Chapter 6 discussed modulation and demodulation, but replaced any detailed discussion of the noise by the assumption that a minimal separation is required between each pair of signal points. This chapter develops the underlying principles needed to understand noise, and Chapter 8 shows how to use these principles in detecting signals in the presence of noise.

Noise is usually the fundamental limitation for communication over physical channels. This can be seen intuitively by accepting for the moment that different possible transmitted waveforms must have a difference of some minimum energy to overcome the noise. This difference reflects back to a required distance between signal points, which, along with a transmitted power constraint, limits the number of bits per signal that can be transmitted.

The transmission rate in bits per second is then limited by the product of the number of bits per signal times the number of signals per second, i.e. the number of degrees of freedom per second that signals can occupy. This intuitive view is substantially correct, but must be understood at a deeper level, which will come from a probabilistic model of the noise.

This chapter and the next will adopt the assumption that the channel output waveform has the form y(t) = x(t) + z(t), where x(t) is the channel input and z(t) is the noise.

In Chapter 4, we showed that any ℒ2 function u(t) can be expanded in various orthogonal expansions, using such sets of orthogonal functions as the T-spaced truncated sinusoids or the sinc-weighted sinusoids. Thus u(t) may be specified (up to ℒ2-equivalence) by a countably infinite sequence such as {uk,m; −∞ < k, m < ∞} of coefficients in such an expansion.

In engineering, n-tuples of numbers are often referred to as vectors, and the use of vector notation is very helpful in manipulating these n-tuples. The collection of n-tuples of real numbers is called ℝn and that of complex numbers ℂn. It turns out that the most important properties of these n-tuples also apply to countably infinite sequences of real or complex numbers. It should not be surprising, after the results of the previous chapters, that these properties also apply to ℒ2 waveforms.

A vector space is essentially a collection of objects (such as the collection of real n-tuples) along with a set of rules for manipulating those objects. There is a set of axioms describing precisely how these objects and rules work. Any properties that follow from those axioms must then apply to any vector space, i.e. any set of objects satisfying those axioms. These axioms are satisfied by ℝn and ℂn, and we will soon see that they are also satisfied by the class of countable sequences and the class of ℒ2 waveforms.

Introduction

Digital modulation (or channel encoding) is the process of converting an input sequence of bits into a waveform suitable for transmission over a communication channel. Demodulation (channel decoding) is the corresponding process at the receiver of converting the received waveform into a (perhaps noisy) replica of the input bit sequence. Chapter 1 discussed the reasons for using a bit sequence as the interface between an arbitrary source and an arbitrary channel, and Chapters 2 and 3 discussed how to encode the source output into a bit sequence.

Chapters 4 and 5 developed the signal-space view of waveforms. As explained in those chapters, the source and channel waveforms of interest can be represented as real or complex ℒ2 vectors. Any such vector can be viewed as a conventional function of time, x(t). Given an orthonormal basis {ϕ1(t), ϕ2(t), …} of ℒ2, any such x(t) can be represented as

Each xj in (6.1) can be uniquely calculated from x(t), and the above series converges in ℒ2 to x(t). Moreover, starting from any sequence satisfying Σj|xj|2 < ∞, there is an ℒ2 function x(t) satisfying (6.1) with ℒ2-convergence. This provides a simple and generic way of going back and forth between functions of time and sequences of numbers.

Introduction

This chapter provides a brief treatment of wireless digital communication systems. More extensive treatments are found in many texts, particularly Tse and Viswanath (2005) and Goldsmith (2005). As the name suggests, wireless systems operate via transmission through space rather than through a wired connection. This has the advantage of allowing users to make and receive calls almost anywhere, including while in motion. Wireless communication is sometimes called mobile communication, since many of the new technical issues arise from motion of the transmitter or receiver.

There are two major new problems to be addressed in wireless that do not arise with wires. The first is that the communication channel often varies with time. The second is that there is often interference between multiple users. In previous chapters, modulation and coding techniques have been viewed as ways to combat the noise on communication channels. In wireless systems, these techniques must also combat time-variation and interference. This will cause major changes both in the modeling of the channel and the type of modulation and coding.

Wireless communication, despite the hype of the popular press, is a field that has been around for over 100 years, starting around 1897 with Marconi's successful demonstrations of wireless telegraphy. By 1901, radio reception across the Atlantic Ocean had been established, illustrating that rapid progress in technology has also been around for quite a while.

Digital communication is an enormous and rapidly growing industry, roughly comparable in size to the computer industry. The objective of this text is to study those aspects of digital communication systems that are unique. That is, rather than focusing on hardware and software for these systems (which is much like that in many other fields), we focus on the fundamental system aspects of modern digital communication.

Digital communication is a field in which theoretical ideas have had an unusually powerful impact on system design and practice. The basis of the theory was developed in 1948 by Claude Shannon, and is called information theory. For the first 25 years or so of its existence, information theory served as a rich source of academic research problems and as a tantalizing suggestion that communication systems could be made more efficient and more reliable by using these approaches. Other than small experiments and a few highly specialized military systems, the theory had little interaction with practice. By the mid 1970s, however, mainstream systems using information-theoretic ideas began to be widely implemented. The first reason for this was the increasing number of engineers who understood both information theory and communication system practice. The second reason was that the low cost and increasing processing power of digital hardware made it possible to implement the sophisticated algorithms suggested by information theory.