Review of course EE320 Communications Principles as a Prerequisite for course EE322

1. Qassim UniversityCollege of Engineering Electrical Engineering DepartmentElectronics and Communications Review of course EE320 Communications Principles as a Prerequisite for course EE322 Associate Prof. Dr. Ahmed Abdelwahab

2. Communication System

3. Information Signals The signals emitted by information sources and the signals sent over a transmission channel can be classified into two distinct categories according to their physical characteristics. These two categories encompass analog and digital signals. An analog signalconveys information through a continuous and smooth variation in time of a physical quantity such as optical, electrical, or acoustical intensities and frequencies. Well-known analog signals include audio (sound) and video messages. For example, an electric signal can vary in frequency (such as the kHz, MHz, GHz designations in radio communications), and its intensity can range from low to high voltages. As the signal travels through the channel, various imperfect properties of the channel induce impairments to the signal. These include electrical noise effects, signal distortions, and signal attenuation. The function of the receiver is to extract the weakened and distorted signal from the channel, amplify it, and restore it as closely as possible to its original signal before transmission and passing it on to the message destination.

4. Sources of information: Text, speech, audio, pictures, video and computer data. They are one, two or three dimensions. Information production and Human perceptual system for audio and video. • Communications networks: telephone networks and Computer networks. • Communications channels: guided propagation such as: telephone channels (two-wire, coaxial cable or optical fiber) and free propagation (wireless channels) such as: broadcasting channels, mobile channels and satellite channels

5. The Modulation Process The purpose of a communication system is to deliver a message (information or baseband) signal from an information source in recognizable form to a user destination, with the source and the user being physically separated from each other. To do this, the transmitter modifies the message signal into a form called passband signal suitable for transmission over the channel. This modification is achieved by means of a process known as modulation, which involves varying some parameter of a carrier wave in accordance with the message signal. The receiver reconstructs as closely as possible the original message signal. This reconstruction is accomplished at the receiver by using a process known as demodulation, which is the reverse of the modulation process done at the transmitter.

6. Types of Modulation We may classify the modulation process into continuous-wave modulation and pulse modulation. In continuous-wave (CW) modulation, a sinusoidal wave is used as the carrier. When the amplitude of the carrier is varied in accordance with the message signal, we have amplitude modulation (AM), and when the angle of the carrier is varied, we have angle modulation. The latter form of CW modulation may be further subdivided into: frequency modulation (FM) and phase modulation (PM), in which the instantaneous frequency and phase of the carrier, respectively, are varied in accordance with the message signal. In pulse modulation, on the other hand, the carrier consists of a periodic sequence of rectangular pulses. However, owing to the unavoidable presence of noise and distortion in the received signal, we find that the receiver cannot reconstruct the original message signal exactly. The resulting degradation in overall system performance is influenced by the type of modulation scheme used. Specifically, we find that some modulation schemes are less sensitive to the effects of noise and distortion than others.

7. Continuous-Wave (CW) Modulation

8. Why do we need modulation? • For use of practical antenna size where the length of the antenna is proportional to the signal wavelength(λ) about 1/10 of λ. Therefore, the baseband information signal spectrum needed to be translated up to a higher frequency band by means of modulations for a smaller antenna size. • For use of Multiplexing that is the process of combiningseveral independent message signals for their simultaneous transmisson over the same channel. Multiplexing could be FDM, TDM or CDM or (WDM for optical fibers). • For use of Better signal-to-noise ratio, it is found that some modulation schemes are less sensitive to the effects of noise anddistortion than others.

10. Quadrature Amplitude Modulation • This scheme enables two DSB-SC modulated waves (resulting from the application of two physically independent message signals) to occupy the same channel bandwidth, and then it allows for the separation of the two message signals at the receiver output. It is therefore a bandwidth-conservation scheme. • To maintain synchronization between transmitter and receiver, a pilot signal outside the passband of the modulated signal may be sent . In this method, the pilot signal typically consists of a low-power sinusoidal tone whose frequency and phase are related to the carrier wave c(t); at the receiver, the pilot signal is extracted by means of a suitably tuned circuit and then translated to the correct frequency for use in the coherent detector.

11. Frequency Translation A modulated wave s1 (t) whose spectrum is centered on a carrier frequency f1 , and the requirement is to translate it upward in frequency such that its carrier frequency is changed from f1 to a new value f2. This requirement may be accomplished using the mixer shown in Figure 2.16. Specifically, the mixer is a device that consists of a product modulator followed by a band-pass filter.

12. Multiplexing is another important signal processing operation, whereby a number of independent signals can be combined into a composite signal suitable for transmission over a common channel. Voice frequencies transmitted over telephone systems, range from 300 to 3100 Hz. To transmit a number of these signals over the same channel, the signals must be kept apart so that they do not interfere with each other, and thus they can be separated at the receiving end. This is accomplished by separating the signals either in frequency or in time. The technique of separating the signals in frequency is referred to as frequency-division multiplexing (FDM), whereas the technique of separating the signals in time is called time-division multiplexing (TDM).

13. 2.6 -Angle Modulation Angle modulation in which the angle of the carrier wave is varied according to the baseband information signal. In this method of modulation, the amplitude of the carrier wave is maintained constant. An important feature of angle modulation is that it can provide better discrimination against noise and interference than amplitude modulation. However, this improvement in performance is achieved at the expense of increased transmission bandwidth; that is, angle modulation provides us with a practical means of exchanging channel bandwidth for improved noise performance. Such a tradeoff is not possible with amplitude modulation, regardless of its form.

14. Pulse Modulation In continuous-wave (CW) modulation, some parameter of a sinusoidal carrier wave is varied continuously in accordance with the message signal. In pulse modulation, some parameter of a pulse train is varied in accordance with the message signal. There are two families of pulse modulation: analog pulse modulation and digital pulse modulation. In analog pulse modulation, a periodic pulse train is used as the carrier wave, and some characteristic feature of each pulse (e.g., amplitude, duration, or position) is varied in a continuous manner in accordance with the corresponding sample value of the message signal. Thus in analog pulse modulation, information is transmitted basically in analog form, but the transmission takes place at discrete times. In digital pulse modulation, on the other hand, the message signal is represented in a form that is discrete in both time and amplitude, thereby permitting its transmission in digital form as a sequence of coded pulses; this form of signal transmission has no CW counterpart. The use of coded pulses for the transmission of analog information-bearing signals represents a basic ingredient in the application of digital communications. This chapter may therefore be viewed as a transition from analog to digital communications in study of the principles of communication systems. We begin the discussion by describing the sampling process, which is basic to all pulse modulation systems, whether they are analog or digital.

15. The sampling process The sampling process is usually described in the time domain. Through use of the sampling process, an analog signal is converted into a corresponding sequence of samples that are usually spaced uniformly in time. Clearly, for such a procedure to have practical utility, it is necessary that we choose the sampling rate properly, so that the sequence of samples uniquely defines the original analog signal. This is the essence of the sampling theorem, which is derived in what follows.

16. The Sampling Theorem The sampling theorem for strictly band-limited signals of finite energy may be stated in two equivalent parts, which apply to the transmitter and the receiver of a pulse modulation system, respectively: A band-limited signal of finite energy, which has no frequency components higher than B Hertz, is completely described by specifying the values of the signal at instants of time separated by (1/2B) seconds. A band-limited signal of finite energy, which has no frequency components higher than B Hertz, may be completely recovered from a knowledge of its samples taken at the rate of 2B samples per second. The sampling rate of 2 B samples per second, for a signal whose bandwidth of B Hertz, is called the Nyquist rate; its reciprocal 1/2 B (measured in seconds) is called the Nyquist interval.

17. some aliasing is produced by the sampling process if the sampling frequency is less than Nyquist rate. Aliasing refers to the phenomenon of a high-frequency component in the spectrum of the signal seemingly taking on the identity of a lower frequency in the spectrum of its sampled version, as illustrated in Figure 3.3. To combat the effects of aliasing in practice, we may use two corrective measures, as described here: 1. Prior to sampling, a low-pass anti-aliasing filter is used to attenuate those high frequency components of the signal that are not essential to the information being conveyed by the signal. 2. The filtered signal is sampled at a rate slightly higher than the Nyquist rate.

18. The reconstruction filter is a low-pass filter with • A passband extending from - W to W, which is itself determined by the anti-aliasing filter. • A transition band extending (for positive frequencies) from W to fs - W, where fsis the sampling rate. The fact that the reconstruction filter has a well-defined transition band means that it is physically realizable.

19. In pulse-amplitude modulation (PAM), the amplitudes of regularly spaced pulses are varied in proportion to the corresponding sample values of a continuous message signal; the pulses can be of a rectangular form or some other appropriate shape. The dashed curve in fig.3.5 depicts the waveform of a message signal m(t), and the sequence of amplitude-modulated rectangular pulses shown as solid lines represents the corresponding PAM signal s(t). Pulse Amplitude Modulation (PAM)

20. In digital circuit technology, two operations that are jointly called "sample and hold" involve in the generation of the PAM signal. One important reason for intentionally lengthening the duration of each sample is to avoid the use of an excessive channel bandwidth, since bandwidth is inversely proportional to pulse duration T. However, care has to be exercised in how long we make the sample duration T. In order to recover (reconstruct) the original message signal m(t), The PAM signal s(t) is passed through a LPF whose frequency response is defined in Figure 3.4c followed by an equalizer in order to compensate for the amplitude distortion. However, for a duty cycle T/Ts≤ 0.1, the amplitude distortion is less than 0.5 percent, in which case the need for equalization may be omitted altogether.

21. Time Division Multiplexing (TDM) An important feature of the sampling process is a conservation of time. That is, the transmission of the message samples engages the communication channel for only a fraction of the sampling interval on a periodic basis, and in this way some of the time interval between adjacent samples is cleared for use by other independent message sources on a time-shared basis resulting in a time-division multiplex (TDM) system, which enables the joint utilization of a common communication channel by a plurality of independent message sources without mutual interference among them.

22. Synchronization is essential for a satisfactory operation of the TDM system. The way this synchronization is implemented depends naturally on the method of pulse modulation used to transmit the multiplexed sequence of samples. The TDM system is highly sensitive to dispersion in the common channel, that is, to variations of amplitude with frequency or lack of proportionality of phase with frequency. Accordingly, accurate equalization of both magnitude and phase responses of the channel is necessary to ensure a satisfactory operation of the TDM system TDM is immune to nonlinearities in the channel as a source of cross-talk. Because different message signals are not simultaneously applied to the channel.

23. The Quantization Process The sampling process takes care of the discrete-time representation of the message signal while the quantization process takes care of the discrete amplitude representation of the message signal. A continuous signal, such as voice, has a continuous range of amplitudes and therefore its samples have a continuous amplitude range. In other words, within the finite amplitude range of the signal, we find an infinite number of amplitude levels. It is not necessary in fact to transmit the exact amplitudes of the samples. Any human sense (the ear or the eye), as ultimate receiver, can detect only finite intensity differences. This means that the original continuous signal may be approximated by a signal constructed of discrete amplitudes selected on a minimum error basis from an available set.

24. The amplitude quantization is defined as the process of transforming the sample amplitude m(nTs) of a message signal m(t) at time t = nTs into a discrete amplitude v(nTs) taken from a finite set of possible amplitudes. the quantization process is assumed memoryless and instantaneous, which means that the transformation at time t = nTs is not affected by earlier or later samples of the message signal. This simple form of scalar quantization, though not optimum, is commonly used in practice. The signal amplitude m is specified by the index k if it lies inside the partition cell where L is the total number of amplitude levels used in the quantizer. Note that the notation is simplified by dropping the time index.

25. Pulse Code Modulation (PCM) In pulse-code modulation (PCM), a message signal is represented in discrete form in both time and amplitude. This form of signal representation permits the transmission of the message signal as a sequence of coded binary pulses. Given such a sequence, the effect of channel noise at the receiver output can be reduced to a negligible level simply by making the average power of the transmitted binary PCM wave large enough compared to the average power of the noise.

27. Binary Encoding To exploit the advantages of sampling and quantizing for the purpose of making the transmitted signal more robust to noise, interference and other channel impairments, we require the use of an encoding process to translate the discrete set of sample values to a more appropriate form of signal. A particular arrangement of symbols used in a code to represent a single value of the discrete set is called a codeword or character. The two symbols of a binary code are customarily denoted as 0 and 1. in a binary code, each codeword consists of R bits. Thus R denotes the number of bits per sample. Then, using such a code, we may represent a total of 2R distinct samples. For example, a sample quantized into one of 256 levels may be represented by an 8-bit codeword.

28. Line Coding (a) In unipolar NRZ signaling, symbol 1 is represented by transmitting a pulse of amplitude A for the duration of the symbol, and symbol 0 is represented by switching off the pulse (on-off signaling). Disadvantages of on-off signaling are the waste of power due to the transmitted DC level and the fact that the power spectrum of the transmitted signal does not approach zero at zero frequency. (b) in Polar NRZ signaling, symbols 1 and 0 are represented by transmitting pulses of amplitudes +A and -A, respectively. This line code is relatively easy to generate but its disadvantage is that the power spectrum of the signal is large near zero frequency.

29. (c) in unipolar RZ signaling, symbol 1 is represented by a rectangular pulse of amplitude A and half-symbol width, and symbol 0 is represented by transmitting no pulse. An attractive feature of this line code is the presence of delta functions at f = 0, ±l/Tb in the power spectrum of the transmitted signal, which can be used for bit timing recovery at the receiver. However, its disadvantage is that it requires 3 dB more power than polar return-to-zero signaling for the same probability of symbol error (d) in bipolar RZ signaling, three amplitude levels are used. Specifically, positive and negative pulses of equal amplitude (i.e., +A and -A) are used alternately for symbol 1, with each pulse having a half-symbol width; no pulse is always used for symbol 0. A useful property of the BRZ signaling is that the power spectrum of the transmitted signal has no DC component and relatively insignificant low-frequency components for the case when symbols 1 and 0 occur with equal probability. (e) Split-phase (Manchester code) In this method of signaling, symbol 1 is represented by a positive pulse of amplitude A followed by a negative pulse of amplitude -A, with both pulses being half-symbol wide. For symbol 0, the polarities of these two pulses are reversed. The Manchester code suppresses the DC component and has relatively insignificant low-frequency components, regardless of the signal statistics. This property is essential in some applications.

30. Regenerative Repeater The most important feature of PCM systems lies in the ability to control the effects of distortion and noise produced by transmitting a PCM signal through a channel. This capability is accomplished by reconstructing the PCM signal by means of a chain of regenerative repeaters located at sufficiently close spacing along the transmission route. The equalizer shapes the received pulses so as to compensate for the effects of amplitude and phase distortions produced by the no ideal channel. The timing circuitry provides a periodic pulse train, derived from the received pulses, for sampling the equalized pulses at the instants at time where the signal-to-noise ratio is a maximum. Each sample so extracted is compared to a predetermined threshold in the decision-making device. In each bit interval, a decision is then made whether the received symbol is a 1 or a 0 on the basis of whether the threshold is exceeded or not. If the threshold is exceeded, a clean new pulse representing symbol 1 is transmitted to the next repeater. Otherwise, another clean new pulse representing symbol 0 is transmitted.

31. In practice, however, the regenerated signal departs from the original signal for two main reasons: The unavoidable presence of channel noise and interference causes the repeater to make wrong decisions occasionally, thereby introducing bit errors into the regenerated signal. If the spacing between received pulses deviates from its assigned value, a jitter is introduced into the regenerated pulse position, thereby causing distortion.

32. PCM Receiver Decoding: The first operation in the receiver is to regenerate (i.e., reshape and clean up) the received pulses one last time. These clean pulses are then regrouped into code words and decoded (i.e., mapped back) into a quantized PAM signal. Filtering: The final operation in the receiver is to recover the message signal by passing the decoder output through a low-pass reconstruction filter whose cutoff frequency is equal to the message bandwidth B. Assuming that the transmission path is error free, the recovered signal includes no noise with the exception of the initial distortion introduced by the quantization process.

33. Noise Considerations in PCM Systems The performance of a PCM system is influenced by two major sources of noise: 1. Channel noise, which is introduced anywhere between the transmitter output and the receiver input. Channel noise is always present, once the equipment is switched on. 2. Quantization noise, which is introduced in the transmitter. Unlike channel noise, quantization noise is signal dependent in the sense that it disappears when the message signal is switched off.

34. Advantages of PCM In a generic sense, pulse-code modulation (PCM) has emerged as the most favored modulation scheme for the transmission of analog information-bearing signals such as voice and video signals. We may summarize the important advantages of PCM as follows: 1. Robustness to channel noise and interference. 2. Efficient regeneration of the coded signal along the transmission path. 3. Efficient exchange of increased channel bandwidth for improved signal-to- noise ratio, obeying an exponential law. 4. A uniform format for the transmission of different kinds of baseband signals, hence their integration with other forms of digital data in a common network. 5. Comparative easewith which message sources may be dropped or reinserted in a time-division multiplex system. 6. Secure communication through the use of special modulation schemes or encryption. These advantages, however, are attained at the cost of increased system complexity and increased channel bandwidth.

35. T1 Carrier System The T1 system, which carries 24 voice channels over separate pairs of wires with regenerative repeaters spaced at approximately 2-km intervals. each of the 24 voice channels uses a binary code with an 8-bit word. The first bit indicates whether the input voice sample is positive (1) or negative (0). The next three bits of the code word identify a particular segment inside which the amplitude of the input voice sample lies, and the last four bits identify the actual representation level inside that segment. With a sampling rate of 8 kHz,each frame of the multiplexed signal occupies a period of 125 μsec.In particular, it consists of twenty-four 8-bit words, plus a single bit that is added at the end of the frame for the purpose of synchronization. Hence, each frame consists of a total of (24 x 8) + 1 = 193 bits. Correspondingly, the duration of each bit equals 0.647 μsec, and the resulting transmission rate is 1.544 megabits per second (Mb/s).

36. Time Division Multiplexing (TDM) The T1 carrier (1.544 Mbps)

37. TDM HierarchyMultiplexing T1 streams into higher carriers T1 (DS1) consists of 24 Telephone calls T2 (DS2) consists of 24*4= 96 Telephone calls T3 (DS3) consists of 96*7=672 Telephone calls T4 (DS4) consists of 672*6= 4032 Telephone calls