First Edition : 2007 - 2008 Communications J. S. Chitode Technical Publications Pune"g i Digital Communications ISBN 978 - 81-8431. 277-5 All rights reserved with Techrical Publications. No port of this book should be reproduced in any form, Electronic, Mechanical, Photocopy or any information storage and reiieval system without prior permission in writing, from Technical Publications, Pune. Published by : Technical Publications Pune” #1, Amit Residency, 412, Shaniwar Peth, Pune - 411 030, Ina Printer : Abet DTPintes Seno. 10/3 Snhased Read, Pine = 411 041Table of Contents 1.1 Advantages of Digital Communication System.... 1.2 Elements of Digital Communication System...... 1.3 Sampling Process 1.3.1 Representation of CT Signals by its Samples . 4.3.2 Sampling Theorem for Lowpass (LP) Signals . 1.3.3 Effects of Undersampling (Aliasing) 1.3.4 Nyquist Rate and Nyquist Interval 1.3.5 Reconstruction Filter (Interpolation Filter) 1.3.7 Sampling Theorem in Frequency Domain ..........--- se sess eeeeeeeeseeeeeee 41.3.8 Sampling of Bandpass Signals 1.4 Pulse Amplitude Modulation (PAM) 4.4.4 Ideal Sampling or Instantaneous Sampling or Impulse Sampling . 1.4.2 Natural Sampling or Chopper Sampling .. 1.4.3 Flat Top Sampling or Rectangular Pulse ee 1.4.4 Comparison of Various Sampling Techniques 1.4.5 Transmission Bandwidth of PAM Signal . 1.4.6 Disadvantages of PAM 1.5 Other Forms of Pulse Modulation. 1-47 1.5.1 Generation of PPMandPDM... wl OB 52: Jon Bandwidth of PPM and PD 49 4.5.3 Comparison between Various Pulse Modulation Methods .............:.ssss00 41-491.6 Bandwidth Noise Trade-off 1.7 Time Division Multiplexing (PAM/TDM System) 1.7.1 Block Diagram of PAM / TDM. 4.7.2 Synchronization in TDM System 1.8.4 Uniform Quantization (Linear Quantization)... ......0:.ssessesseeeeeseeeeeere 1-66 18.4.1 Midtread Quanfizer, 0 fy tied Cadi st B43 sen Cece 1.8.5 Quantization Noise and Signal to Noise Ratio in PCM. ...........sseeseseeeeere 4-70 1,8.5.1 Derivation of Quantization Error/Noise or Noise Power for Uniform (Linear) Quantization 1 - 70 1.8.5.2 Derivation of Maximum Signal to Quantization Noise Ratio for Linear Quantization. ._ 1-73 1864p. ‘Law Companding for Speech Signals. SoA siecacar a neaee 1.8.6.5 ALawfor Companding. .... .. .. imipenem pote 4.8.6.8 Signal to Noise Ratio of Companded PCM... 1.9 Digital Multiplexers... 1.9.1 Types of Digital Multiplexers 1.92 Multiplexing Hierarchies .. 1.93 PCM TOM System... 1.9.3.1 Multiplexing Hierarchy = = ee .2 Multiple Channel Frame Alignment For TDM/PCM (T,System) . . . . wana 1.10 Virtues, Limitation and Modifications of PCM ........ ele leis 1.11.2 Principle of DPCM... sssssssessseee siasie satan’ swileebg 1-105 1.11.4 Reconstruction of DPCM Signal: . 21-106 221 Dials MOU UHR csicccnrenmerenama cnn RS SRE EO2.1.4 Operating Principle of DM ..........cecsseeeseeesee eee eeesieesa sees eeseees 2-1 G2 EAR STR RR a ee 2.2 Advantages and Disadvantages of Delta Modulation .. 2.2.1 Advantages of Delta Modulation, . 2.22 Disadvantages of Delta Modulation .. 2.2.2.4 Slope Overload Distortion (Startup Error) . 2.2.22 Granular Noise (Hunting) . 2.3 Adaptive Delta Modulation .. 2.3.4 Operating Principle... mt 2.3.3 Advantages of Adaptive Delta Modulation . . 2.4 Comparison of Digital Pulse Modulation Method: 3.1.4 Types of Passband Modulation............... 3.1.2 Types of Reception for Passband Transmission 3.1.3 Requirements of Passband Transmission Scheme. 3.1.4 Advantages of Passband Transmission over Baseband Transmission. 3.2 Binary Phase Shift Keying (BPSK) 3.211 Principle of BPSK. 5.22 Graphical Representaion of BPSK Signal 3.2.3 Generation and Reception of BPSK Signal. 4.2.3.1 Generator of BPSK Signal. . 3.2.32 Reception of BPSK Signal. 3.24 Spectum of BPSK Sina. 3.2.6 Bandwidth of BPSK Signal .. 3.2.7 Drawbacks of BPSK : Ambiguity in Output Signal. 3.3 Differential Phase Shift Keying (DPSK)..... 3.4 aaa Phase Shift Keying ....3.4.1 QPSK Transmitter and Receiver 3.4.1.1 Offset QPSK (OQPSK) or Staggered QPSK Transmitter. . mn 3 4.1.4 Carter Synchronization in QPSK... . 3.4.2 Signal Space Representation of QPSK Signals 3.4.3 Spectrum of QPSK Signal 3.4.4 Bandwidth of QPSK Signal 3.4.5 Advantages of QPSK... 3.5.3 Bandwidth of M-ary PSK 3.5.4 Distance between Signal Points (Euclidean Distance) 3.5.5 Transmitter and Receiver of M-ary PSK . 3.5.5.1 M-aty PSK Transmitter... . 3.55.2 Meaty PSKReceiver. . . . 3.6 Quadrature Amplitude Shift Keying (QASK) [or Quadrature Amplitude Modulation (QAM)... 3.6.1 Geometrical Repre sentation and Euclidean Distance of QASK Signals (or Signal Space Representation or Signal Space Constellation)... 3.62.4 Transmiter of QASK Signal for 4-bt Symbol. . 3.6.2.2 Receiver of QASK Signal... . 36.3 Power Spectral Density and Bandwidth of QASK Signal .. 3.6.4 Comparison between QASK and QPSK... 3.7 Binary Frequency Shift Keying (BFSK) 37.7 Advantages and Disadvantages of BESK 3.8 M-ary FS!3.8.3 Geomettical Representation of M-ary FSK or Signal Space Representation 3.9 Minimum Shift Keying (MSK).. 3.9.1 Signal ation of MSK and Distance between the Signal Poin (or Geometrical Representation of MSK). . 3.9.2 Power Spectral Density and Bandwidth of MSK . 3.9.3 Phase Continuity in MSK. 3.10 Amplitude Shift Keying or ON-OFF Keying 3.10.1 Signal Space Diagram of ASK. 3.10.2.3 Noncoherent ASK Reception . 5 3.11 Comparison of Digital Modulation Techs. 4.1 Baseband Signal Receiver... 4.1.1 Signal to Noise Ratio of the Integrator and Dump Fitter . 4.1.2 Probability of Error in Integrate and Dump Filter Receiver 4.2 Optimum Receiver (or Optimum Filter)... 4.2.1 Probabilty of Error of Optimum Filter... 4.2.2 Transfer Function of the Optimum Filter 4.3 Matched Filter .. 4.3.1 Impulse Response rr 4.3.2 Probability of Error of the Matched Filter . . 4.5 Error Probabilities of Baseband Signaling Schemes 4.5.1 Detection of PCM Signal 4.5.2 Error Probability of ASK 4.5.3 Probability of Error for Coherently Detected BPSK .4.5.3.1 Efiect of Imperfect Phase Synchronization on Output and P, 45.3.2 Effect of Imperiact Bit Synchronization on Output andP, 4.54 Probabiliy of Eor for Coherently Detected Binary Orthogonal FSK. . 4.55 Probability of Error for Non-Coherently Detected Binary Orthogonal FSK. 4.56 Probabiliy Error for Binary Orthogonal DPSK . 4.517 Probabilty of Error for QPSK. 4.6 Signal Space to Calculate P.. 4.8.1 Error Probability of BPSK. 4.6.2 Error Probability of BFSK . 5868 F ce Hh 4-66 4-68 STNUOGUGION seca nme aE Oe 5.2 Uncertainty... 5.3 Definition of Information (Measure of Information’ 5.3.1_ Properties of Information 5.3.2 Physical Interpretation of Amount of Information. 5.4 Entropy (Average Information} 5.4.1 Properties of Entropy 5.5 Information Rate ... 5.6 Discrete Memoryless Channels 5.6.1 Binary Communication Channel. Equivocation (Conditional Entropy) 5.64 comscly ‘of a Discrete Memoryless Channel. $7 Mutual (information eee = AB 5.7.1 Properties of Mutual Information 5.7.2 Channel Capacity 5.8 Differential Entropy and Mutual Information for Continuous Ensembles .5 - 69 5.8.1 Difforontial Entropy........++c0ssscssserserecsecenecrcsersoesseseesssores 5-70 6.2 Source Coding Theorem (Shannon's First Theorem). 6.2.1 Code Redundancy.6.3.4 Prefix Coding (Instantaneous Coding) . 6.3.1.1 Properties of Prefix Code . 6.3.2 Shannon-Fano Algorithm... .. 6.3.3 Huffman Coding..................64 6.4 Shannon's Theorems on Channel Capacity 6.4.1 Channel Coding Theorem (Shannon's Second Theorem) . 6.4.2 Shannon Hartley Theorem for Gaussian Channel (Continuous Channel) . 7.4.4 Rationale for Coding and Types and Codes: 7.1.2 Types of Codes 7.1.3 Discrete Memoryless Channels. . 7.1.4 Examples of Error Control Coding. 7.1.5 Methods of Controlling Errors . 7.1.8 Types of Errors... = 7.1.7 Some of the Important Terms used in Error Control Coding 7.2 Linear Block Codes...... 7.2. Matrix Description of Linear Blocks Codes 7.2.2 Hamming Codes 7.2.3 Error Detection and Correction Capabili 7.2.4 Encoder of (7, 4) Hamming Code . . 7.2.5 Syndrome Decoding .......... 7.2.5.1 Error Correction using Syndrome Vector. . 7.2.6 Hamming Bound : 7.2.7 Syndrome Decoder for (n, k) Block Code........sssseseseeseesessessessesee 1.2.8 Offer Lingar Block COdGS ooo eet ete ene OS 7.3 Binary Cyclic Codes... 7.3.1 Definition of Cyclic Code . 7.32 Properties of Cyolic Codes7.3.24 Lineaity Property ©... 0... 73.22 CyollePropery 7.33 Algebraic Structures of Gycho Codes. 7.3.3.1 Generation of Code vectors in Nonsystematic Form 7.3.3.2. Generation of Code vectors in Systematic Form... 7.34 Generator and Parity Check Matrices of Cyclic Codes 7.3.44 Nonsystematic Form of Generator Matrix... 7.3.42 Systematic Form of Generator Matrix . 7.3.43 Parity Check Matrix. 6.6... 7.3.5 Encoding using an (n —k) Bit Shift Register .. 7.3.6 Syndrome Decoding, Error Detection and Error Correction . 7.3.6.1 Biock Diagram of Syndrome Calculator ea 7.3.7 Decoder for Cyclic Codes................ 7.38 Advantages and Disadvantages of Cyclic Codes. . 7.39 BCH Codes (Bose - Chaudhri - Hocquenghem Codes) 7.310 Reed-Soloman (RS) Codes . 7.31 Golay Codes. 7.3.12 Shortened Cyclic Codes. . 7.3.13 Burst Error Correcting Codes 7.3.14 Interleaving of Coded Data for Burst Error Correction siglsisalales a lala lelala a MS AS a 7.3.16 Cyclic Rkacj Ch check (eRC) Codes 7.3.17 Maximum length codes 8.1.2 Code Rate of Convolutional Encoder 8.1.3 Constraint Length (K) 8.1.4 Dimension of the Code 8.2 Analysis of Convolutional Encoder 8.2.1 Time Domain Approach to Analysis of Convolutional Encoder . . 8.22 Transform Domain Approach to Analysis of Convolutional Encoder . 8.3 Code Tree, Trellis and State Diagram for a Convolution Encoder 8.3.1 States of the Encoder. 8.3.2 Development of the Code Tree . 8.33 Code Trelis (Represents Steady State Transion).. squsetenoneeenanemeaenecned8.34 State Diagram. 8.4 Decoding Methods of Convolutional Codes. 8.4.1Vterbi Algorithm for Decoding of Convolutional Codes (Maximum Likelihood Decoding) 8 - 18 8.4.2 Sequential Decoding for Convolutional Codes 28-20 8.4.3 Free Distance and Coding Gain ........ HetTiansieFincien ata = ra 8.6 Distance Properties of Binary Convolutional Codes... 8.7 Advantages and Disadvantages of Convolutional Codes .. 8.8 Comparison between Linear Block Codes and Convolutional Codes.8 - 57ADM Adaptive Delta Modulation FDMA | Frequency Division Multiple Access ASK Amplitude Shift Keying FSK Frequency Shift Keying AWGN | Additive White Gaussian Noise Is Inter Symbol Interference BER Bit Error Rate ISDN Integrated Services Digital Network BPF Bandpass Filter MSK Minimum Shift Keying BSC Binary Symmetric Channel Nonreturn to Zero BW Bandwidth On-off Keying BPSK Binary Phase Shift Keying Pulse Amplitude Modulation BFSK Binary Frequency Shift Keying Pulse Duration Modulation cw Continuous Wave Probability Density Function cDM Code Division Multiplexing Power spectral density CDMA _| Code Division Multiple Access Phase Shift Keying CDF Cumulative Distribution Function Phase Locked Loop B Decibel Pulse Position Modulation DM Delta Modulation Pulse Width Modulation DPCM __| Differential Pulse Code Modulation Quadrature Amplitude Modulation DPSK Differential Phase Shift Keying Quadrature Phase Shift Keying DSB Double Sided Modulation Return to Zero ef Error function ‘Signal to Noise Ratio erfe Complementary error function Signal to Noise Ratio exp Exponential (e) =F apy sense Soul, ‘Time Division Multiplexing 7a | Berag Deaaaamee | (xiv)‘Magnitude of the complex quantity contained within Mean or average value of 2 random variable X ow) Fourier transform pair ox Standard deviation of a random varible X Time average of x(1) a Variance of « random variable X * denctes complex conjugate fancton x | Has Xs commer comes | 95 Energy spectral density of a signal x(t) Convolution of x (0) and y (0 a0 Pivwor special deny x pax Eo as Se hw Impulse response of the linear system aa pat eal and GA HD ‘Transfer function of the linear system imaginary part of the quantity contained Ty Duration of one bit within B or By | Transmission channel bandwidth in F(x (0) _| Fourier transform of x () 8g __| tieoorinin : r Signaling rate in digital tansmission. It Inverse fourier transform of X (f) nent ene fas }——_| - v Number of binary bits wed for IFT (X(6] | Inverse fourier transform of X (6) encoding a sample value ® Modulo - 2 Addition i. Exctuive-oR | |g Number of digital levels used to operation encode a sample value 20) ‘Symbol * represents that (1) is the | | 3 Step size of the quantizer used in reconstructed value of x (\) in receiver Digital Modulation Methods i.e. PCM, DPCM, DM, ADM etc fox} Represents sequence whose k! value is ae R Symbol power EU] Expected value or mean value of the | | J ‘Average interference’ power random vatiable contained within 5 Gascon rer Sale mola Probability of error of symbol or bit methods fy @) | Probability density function of a] | (0 Coherent / noncoberent cartier & continuous random variable at X = x. reference signal used in digital passband transmission Probability density function of random a variables X& YatXex& ¥=y 6 Generator polynomial in eyelie codes Probability of a random variable X for| | X() Code veetor polynomial posi MO) “Message bits polynomial cw Cheek bits polynomialCopyrighted materialPulse Digital Modulation Introduction © There are three types of modulation (Amplitude modulation (i) Angle modulation (iii) Pulse modulation « Pulse modulation can be further classified as, (Pulse analog modulation (i) Pulse digital modulation * The above two techniques can be further classified as, (i) Pulse amplitude modulation (Pulse code modulation (ii) Pulse position modulation (i) Detta modulation (iii) Pulse duration modulation (ii) Adaptive dotta modulation {iv) Differential puise code modulation In the above techniques following points are studied : (i) Principle of operation (ji) Transmitter and receiver block diagram (ii) Exror analysis (iv) Signal to quantization noise ratio. (1-4)Digital Communications 1-2 Pulse Digital Modulation 1.1 Advantages of Digital Communication System Presently most of the communication is digital. For example cellular (mobile phone) communication, satellite communication, radar and sonar signals, Facsimile, data transmission over internet etc all use digital communication. Paractically, after 20 years, analog communication will be totally replaced by digital communication. Why digital communication is so popular ? There are few reasons due to which people are prefering digital communication over analog communication. 1. Due to advancements in VLSI technology, it is possible to manufacture very high speed embedded circuits. Such circuits are used in digital communications. 2. High speed computers and powerful software design tools are available. They make the development of digital communication systems feasible. 3. Intemet is spread almost in every city and towns. The compatibility of digital communication systems with internet has opened new area of applications. Advantages and Disadvantages of Digital Communication Advantages : 1. Because of the advances in digital IC technologies and high speed computers, digital communication systems are simpler and cheaper compared to analog systems. 2. Using data encryption, only permitted receivers can be allowed to detect the transmitted data. This is very useful in military applications. 3. Wide dynamic range is possible since the data is converted to the digital form. 4. Using multiplexing, the speech, video and other data can be merged and transmitted over common channel. 5. Since the transmission is digital and channel encoding is used, the noise does not accumulate from repeater to repeater in long distance communication. 6. Since the transmitted signal is digital, a large amount of noise interference can be tolerated. 7. Since channel coding is used, the errors can be detected and corrected in the receivers. 8. Digital communication is adaptive to other advanced branches of data processing such as digital signal processing, image processing, data compression etc.Digital Communications 1-3 Pulse Digital Modulation Disadvantages : Eventhough digital communication offer many advantages as given above, it has some drawbacks also. But the advantages of digital communication outweigh disadvantages. They are as follows - 1. Because of analog to digital conversion, the data rate becomes high. Hence more transmission bandwidth is required for digital communication. 2. Digital communication needs synchronization in case of synchronous modulation. 1.2 Elements of Digital Communication System Fig. 1.2.1 shows the basic operations in digital communication system. The source and the destination are the two physically separate points. When the signal travels in the communication channel, noise interferes with it. Because of this interference, the smeared or disturbed version of the input signal is received at the receiver. Therefore the signal received may not be correct. That is errors are introduced in the received signal. Thus the effects of noise due to the communication channel limit the rate at which signal can be transmitted. The probability of error in the received signal and transmission rate are normally used as performance measures of the digital communication system. Fig. 1.2.1 Basic diaital communication system 4.2.1 Information Source The information source generates the message signal to be transmitted. In case of analog communication, the information source is analog. In case of digital communication, the information source produces a message signal which is not continuously varying with time. Rather the message signal is intermittent with respect to time. The examples of discrete information sources are data from computers,Digital Communications 1-4 Pulse Digital Modulation teletype etc. Even the message containing text is also discrete. The analog signal can be converted to discrete signal by sampling and quantization. In sampling, the analog signal is chopped off at regular time intervals. Those chopped samples form a discrete signal. The discrete information sources have following important parameters : a) Source alphabet : These are the letters, digits or special characters available from the information source. 'b) Symbol rate : It is the rate at which the information source generates source alphabets. It is normally represented in symbols/sec unit. © Source alphabet probabilities : Each source alphabet from the source has independent occurrence rate in the sequence. For example, letters A, E, I etc. occur frequently in the sequence. Thus probability of the occurrence of each source alphabet can become one of the important property which is useful in digital communication. dq) Probabilistic dependence of symbols in a sequence : The information carrying capacity of each source alphabet is different in a particular sequence. This parameter defines average information content of the symbols. The entropy of a source refers to the average information content per symbol in long messages. Entropy is defined in terms of bits per symbol. Bit is the abbreviation for binary digit. The source information rate is thus the product of symbol rate and source entropy i.e.. Information rate = Symbol rate x Source entropy (Bits/sec) (Symbols/sec) _(Bits/Symbol) The information rate represents minimum average data rate required to transmit information from source to the destination. 1.2.2 Source Encoder and Decoder The symbols produced by the information source are given to the source encoder. ‘These symbols cannot be transmitted directly. They are first converted into digital form (i.e. Binary sequence of 1's and 0’s) by the source encoder. Every binary ‘1’ and “0! is called a bit. The group of bits is called a codeword. The source encoder assigns codewords to the symbols. For every distinct symbol there is a unique codeword. The codeword can be of 4, 8, 16 or 32 bits length. As the number of bits are increased in each codeword, the symbols that can be represented are increased. For example, 8 bits will have 2° = 256 distinct codewords. Therefore 8 bits can be used to represent 256 symbols, 16 bits can represent 2'° = 65536 symbols and so on. In both of the above examples the number of bits in every codeword is same throughout. That is 8 in first case and 16 in next case respectively. This is called fixed length coding. Fixed length coding is efficient only if all the symbols occur with equalDigital Communicatio: 1-5 Pulse | Modulation Probabilities in a statistically independent sequence. In the practical situations, the symbols in the sequence are statistically dependent and they have unequal probabilities of occurrence. For example, let us assume that the symbol sequence represents the percentage marks of the students. The 02%, 08%, 20%, 98%, 99% etc. symbols will have minimum probability of occurrence. But 60%, 55%, 70%, 75% will have more probability. For such symbols normally variable length codewords are assigned. More bits (More length) are assigned to rarely occurring symbols and less bits are assigned to frequently occurring symbols. Typical source encoders are pulse code modulators, delta modulators, vector quantizers etc. We will come across these codewords in detail in the subsequent chapters. Source encoders have following important parameters. a) Block size : This gives the maximum number of distinct codewords that can be represented by the source encoder. It depends upon maximum number of bits in the codeword. For example, the block size of 8 bits source encoder will have 28 =256 codewords. Codeword length : This is the number of bits used to represent each codeword. For example, if 8 bits are assigned to every codeword, then codeword length is 8 bits. ©) Average data rate : It is the output bits per second from the source encoder. The source encoder assigns multiple number of bits to every input symbol. ‘Therefore the data rate is normally higher than the symbol rate. For example let us consider that the symbols are given to the source encoder at the rate of 10 symbols/sec and the length of codeword is 8 bits. Then the output data rate from the source encoder will be, Date rate = Symbol rate x Codeword length = 10 x 8 = 80 bits/sec b) Information rate is the minimum number of bits per second needed to convey information from source to destination as stated earlier. Therefore optimum data rate is equal to information rate. But because of practical limitations, designing such source encoder is difficult. Hence average data rate is higher than information rate and hence symbol rate also. d) Efficiency of the encoder : This is the ratio of minimum source information rate to the actual output data rate of the source encoder. At the receiver, some decoder is used to perform the reverse operation to that of source encoder. It converts the binary output of the channel decoder into a symbol sequence. Both variable length and fixed length decoders are possible. Some decoders use memory to store codewords. The decoders and encoders can be synchronous or asynchronous.Digital Communications 1-6 Pulse Digital Modulation 1.2.3 Channel Encoder and Decoder At this stage we know that the message or information signal is converted in the form of binary sequence (ie, 1’s and 0's). The communication channel adds noise and interference to the signal being transmitted. Therefore errors are introduced in the binary sequence received at the receiver. Hence errors are also introduced in the symbols generated from these binary codewords. To avoid these errors, channel coding is done. The channel encoder adds some redundant binary bits to the input sequence: These redundant bits are added with some properly defined logic. For example consider that the codeword from the source encoder is three bits long and one redundant bit is added to make it 4-bit long. This 4" bit is added (either 1 or 0) such that number of 1’s in the encoded word remain even (also called even parity). Following table gives output of source encoder, the 4" bit depending upon the parity, and output of channel encoder. Output of source | Bit to be added by channel | Output of channel encoder encoder for even parity encoder by be by by by by by Table 1.2.1 Even parity coding Observe in the above table that every codeword at the output of channel encoder contains “even” number of 1’s. At the receiver, if odd number of 1's are detected, then receiver comes to know that there is an error in the received signal. The channel decoder at the receiver is thus able to detect error in the bit sequence, and reduce the effects of channel noise and distortion. The channel encoder and decoder thus serve to increase the reliability of the received signal. The extra bits which are added by the channel encoders carry no information, rather, they are used by the channel decoder to detect and correct errors if any. These error correcting bits may be added recurtently after the block of few symbols or added in every symbol as shown in Table 1.2.1. The example of parity coding given above is just illustrative. There are many advanced and efficient coding techniques available. We will discuss them in the book. The coding and decoding operation at encoder and decoder needs the memory (storage) and processing of binary data. Because of microcontrollers and computers, the complexity of encoders and decoders is nowadays very much reduced. The important parameters for channel encoder are -Digital Communications 1-7 Pulse Digital Modulation a) The method of coding used. b) Coding rate, which depends upon the redundant bits added by the channel encoder. c) Coding efficiency, which is the ratio of data rate at the input to the data rate at the output of encoder. d) Error control capabilities, i.e. detecting and correcting errors e) Feasibility or complexity of the encoder and decoder. The time delay involved in the decoding is also an important parameter for channel decoder. 1.2.4 Digital Modulators and Demodulators Whenever the modulating signal is discrete (ie. binary codewords), then digital modulation techniques are used. The carrier signal used by digital modulators is always continuous sinusoidal wave of high frequency. The digital modulators maps the input binary sequence of 1’s and 0's to analog signal waveforms. If one bit at a time is to be transmitted, then digital modulator signal is s;(f) to transmit binary ‘0’ and s9(f) to transmit binary ‘1’. For example consider the output of digital modulator shown in Fig. 1.2.2. soll) sift) s(t) soft) si(t) s(t) Fig. 1.2.2 Frequency modulated output of a digital modulator The signal s,() has low frequency compared to signal s(t). It is frequency modulation (FM) in two steps corresponding to binary symbols ‘0’ and ‘1’. Thus even though the modulated signal appears to be continuous, the modulation is discrete (or in steps). Single carrier is converted into two waveforms s(t) and s2(t) because of digital modulation. If the codeword contains two bits and they are to be transmitted at a time, then there will be M=2? =4 distinct symbols (or codewords). These four codewords will require four distinct waveforms for transmission. Such modulators are called M-ary modulators. Frequency Shift Keying (FSK), Phase Shift Keying (PSK), Amplitude Shift Keying (ASK), Differential Phase Shift Keying (DPSK), Minimum Shift Keying (MSK) are the examples of various digital modulators. Since these modulators use continuous carrier wave, they are also called digital CW modulators.Digital Communications 1-8 Pulse Digital Modulation In the receiver, the digital demodulator converts the input modulated signal to the sequence of binary bits. The most important parameter for the demodulator is the method of demodulation. The other parameters for the selection of digital modulation method are, a) Probability of symbol or bit error. b) Bandwidth needed to transmit the signal. c) Synchronous or asynchronous method of detection and d) Complexity of implementation. 1.2.5 Communication Channel As we have seen in the preceding sections, the connection between transmitter and receiver is established through communication channel. We have seen that the communication can take place through wirelines, wireless or fiber optic channels, The other media such as optical disks, magnetic tapes and disks etc. can also be called as communication channel, because they can also carry data through them. Every communication channel has got some problems. Following are the common problems associated with the channels : a) Additive noise interference : This noise is generated due to internal solid state devices and resistors etc. used to implement the communication system. Signal attenuation : It occurs due to internal resistance of the channel and fading of the signal. c) Amplitude and phase distortion : The signal is distorted in amplitude and phase because of non-linear characteristics of the channel. Multipath distortion : This distortion occurs mostly in wireless communication channels, Signals coming from different paths tend to interfere with each other. There are two main resources available with the communication channels. These two resources are - b) dq) a) Channel bandwidth ; This is the maximum possible range of frequencies that can be used for transmission. For example, the bandwidth offered by wireline channels is less compared to fiber optic channels. b) Power in the transmitted signal : This is the power that can be put in the signal being transmitted. The effect of noise can be minimized by increasing the power. But this cannot be increased to very high value because of the equipment and other constraints. For example, the power in the wireline channel is limited because of the cables. The power and bandwidth limit the data rate of the communication channel. As we know, the fiber optic channel transports light signals from one place to another just like a metallic wire carriers an electric signal. There is no current or metallic conductor in optical fiber. The optical fiber has following advantages :Digital Communications 1-9 Pulse Digital Modulation a) Very large bandwidths are possible. b) Transmission losses are very small. c) Electromagnetic interference is absent. d) They have small size and weight. e) They offer ruggedness and flexibility. £) Optical fibers are low cost and cheap. Satellites essentially perform wireless communication. Mainly satellites are repeaters. Broad area coverage is the main advantage of satellites. The power requirement is also less, since solar energy is used by satellites. Global communication is very easily possible through satellite channel. The interference on satellite channels is present but it is minimum. Theory Question 1. Explain with neat block diagram the essential and non essential features of a digital communication system. 1.3 Sampling Process 1.3.1 Representation of CT Signals by its Samples Why CT signals are represented by samples ? * ACT signal cannot be processed in the digital processor or computer. * To enable digital transmission of CT signals. Fig. 1.3.1 shows the CT signal and its sampled DT signal. In this figure observe that the CT signal is sampled at t = 0, T,, 2T, 37, ... and so on. T 0. jpop L 4b ab Le Bod Pee Fig. 1.3.1 CT and its DT signalDigital Communications 1-10 Pulse Digital Modulation * Here sampling theorem gives the criteria for spacing 'T,’ between two successive samples. © The samples xg(t) must represent all the information contained in x(t). ‘The sampled signal 1g(t) is called discrete time (DT) signal. It is analyzed with the help of DIFT and z-transform. 1.32 Sampling Theorem for Lowpass (LP) Signals A lowpass or LP signal contains frequencies from 1 Hz to some higher value. Statement of sampling theorem 1) A band limited signal of finite energy, which has no frequency components| higher than W Hertz, is completely described by specifying the values of the| signal at instants of time separated by We seconds and A band limited signal of finite energy, which has no frequency components| higher than W Hertz, may be completely recovered from the knowledge of its| Samples taken at the rate of 2W samples per second. The first part of above statement tells about sampling of the signal and second part tells about reconstruction of the signal. Above statement can be combined and stated alternately as follows : A continuous time signal can be completely represented in its samples and recovered back if the sampling frequency is twice of the highest frequency content of the signal. i.e., fr 2 2W Here f, is the sampling frequency and W is the higher frequency content Proof of sampling theorem There are two parts : (1) Representation of a(t) in terms of its samples (@) Reconstruction of (f) from its samples. Part I: Representation of x(t) in its samples (nT,)Digital Communications 1-11 Pulse Digital Modislation Step 1: Define x5(t) Refer Fig, 1.3.1. The sampled signal xg(t) is given as, xg) = Dx)b(t-n7,) es (13.1) ace Here observe that xg(t) is the product of xs and impulse train 8(() as shown in Fig. 13.1. In the above equation &(!~nT,) indicates the samples placed at T,, +2T,, £37, ... and so on. Step 2: FT of xa(t)ie. X(f Taking FT of equation (1.3.1). = Zensen} xs = FT (Product of x(#) and impulse train} We know that FT of product in time domain becomes convolution in frequency domain. i.e., Xs) = FT ed) *FT(S(t-n7,)) (1.3.2) By definitions, a(t) <7?» x() and Bt-nt) Lag F8G-nf) Hence equation (1.32) becomes, Xs = XN*f Y5(F-nf) nace Since convolution is linear, Xs = & SX 3G-n6) fe SXU-né) By shifting property of impulse friction nae Ff RUE 2 fe) + fe XU — fa) + fe XUV + fe XU ~ Sa) +f MUP 2h)Digital Communications 4-42 Pulse Digital Modulation Comments (i) The RHS of above equation shows that X(f) is placed attf,,t2f,,43f,,° (i) This means X() is periodic in f,. (iii) If sampling frequency is f, = 2W, then the spectrums X(f) just touch each other. Fig, 1.3.2 Spectrum of original signal and sampled signal Step 3 : Relation between X(f) and X5(f) Important assumption : Let us assume that f, = 2W, then as per above diagram. Xs) = £XN for - W $f@, -0,, This filter provides reverse action to that of zero-order hold, Fig, 1.3.10 shows the block diagram with anit-imaging filter. Yelt) x(n) Fig. 1.3.10 Block diagram of practical reconstruction 1.3.7 Sampling Theorem in Frequency Domain Statement We have seen that if the bandlimited signal is sampled at the rate of (f, > 2W) in time domain, then it can be fully recovered from its samples. This is sampling theorem in time domain. A dual of this also exists and it is called sampling theorem in frequency domain. It states that,Digital Communications 1-21 Pulse Digital Modulation signal which is zero for |t|>T is uniquely determined by the samples of at intervals less than o Hertz apart”. + Explanation : Thus the spectrum is sampled at f, <2 in the frequency domain. T is the maximum time limit above which signal x() goes to zero. ‘f,' represents the sampling frequency interval in the frequency spectrum of the signal. Note that here f, does not represent number of samples taken per second. But it represents the frequency interval at which the samples are separated in frequency domain. + Fig. 1.3.11 illustrates the sampling theorem in frequency domain. We can see from 13.11 (a) that a rectangular pulse is time limited to af seconds Le, x)= for -EstW. * Inphase and quadrature components : This bandpass signal is first represented in terms of its inphase and quadrature components. Let x(t) = Inphase component of x() and xQ() = Quadrature component of »(#) Then we can write x(/) in terms of inphase and quadrature components as, ME) = x; (8) cos (2nf,t)-xQ (f) sin (2nft) w+» (13.10) The jinphase and quadrature components are obtained ~— by multiplying x(!) by cos (2nf,t) and sin (2nf,t) and then suppressing the sum frequencies by means of low-pass filters. Thus inphase x, () and woof OW { quadrature Xg (t) components contain Fig. 1.3.13 Spectrum of inphase and only low frequency components. The «(aW7) sine(2t—3) eo [2.('-aw)] ves (13.11) Compare this reconstruction formula with that of lowpass signals given by equation (1.3.6). It is clear that a(#) is represented by x (aw) completely. Here, “() and Te = x (nT) = Sampled version of bandpass signal ak, iw © Thus if 4W samples per second are taken, then the bandpass signal of bandwidth 2W can be completely recovered from its samples. Thus, for bandpass signals of bandwidth 2W, Minimum sampling rate = Twice of bandwidth = 4W samples per second wm Example’ 1.3.1: Show that a bandlimited signal of finite energy which has no frequency components higher than W Hz is completely described by specifying values of the signals at instants of time separated by 1/W seconds and also show that if the instantaneous values of the signal are separated by interoals larger than 1 seconds, they fail to describe the signal. A bandpass signal has spectral range that extends from 20 to 82 kHz, Find the acceptable range of sampling frequency f.. Solution : Step 1: Define xg(t). Let x{#) be the bandlimited signal which has no frequency components higher than W Hz. Let it be sampled by a sampling function x= ¥ 8-07) nse The sampling function is the train of impulses with T, as distance between successive impulses. Let x(nT;) be the instantaneous amplitude of signal x() at instant t=T,. The sampled version of af) can be represented as multiplication of x (nT,) and &() ie.Digital Communications 4-25 Pulse Digital Modulation Xs) = YS x(n, 8(t- nT) e+ (1.3.12) n= Step 2: Fourier transform of xg(t) ie. X5(f) Fourier transform of this sampled signal can be obtained as, Xa) = FI{xs)} Xo = fe > X(f- nf) ws (13.13) ies Here f, is the sampling rate which is given as f, =3 fs And, X (f - nf,)=X(f) at nf, =0, +f,,2f,t3f,.... Thus the same spectrum X(f) appears at f=0, f=+/,,f=+2f, etc. This means that a periodic spectrum with period equal to f, is generated in frequency domain because of sampling x() in time domain. Therefore equation 1.3.13 can be written as, Xa = fe XN+h Xft) +h XP E25) FAX PARA +A XP E46) w+ (13.14) or Xa) = K$XN+ YK X-nf) o+- (13.15) nase ned Step 3: Relation between X(f) and X5(/). By definition of Fourier transform, X(f)= [ xf) e/?™" dt For sampled version of x(t), we have t= nT,. Then above equation becomes, Xs) = Y xT, ye Pas +++ (13.16) nase It is given that the signal is band limited to W Hz and, = : eit T, = gy seconds, «fsa q-=2W w+ (13.17) From equation 13.14 we know that Xs (/) is periodic in f,. The spectrum X(f) and Xs (f) are shown in Fig, 1.3.14.Digital Communications 1-26 Pulse Digital Modulation (a) +f s , TW 2h " =-3w =+3W Fig. 1.3.14 (a) Spectrum of x(t) & Spectrum oF ete) with f,=2wW Since f,=2W; f,-W=W and f,+W=3W Thus the periodic spectrums X(f) just touch +W, £3W , t5W.... ete. Thus there is no aliasing. From equation 1.3.15. we can write, ji we XP) = FXo- ¥ XG-ne) ss (13:18) - n=-@ nad With f, = 2W in above equation, 1 « XO = ayX~_% XU-m1) 00 ie xp = xi For -W, X-f,) we (1.4.3) Comments i) X()is periodic in f, and weighed by f,. ii) Instantaneous sampling is possible only in theory because it is not possible to have a pulse whose width approaches zero 1.4.2 Natural Sampling or Chopper Sampling * Basic Principle . In natural sampling the pulse has a finite width t. Natural sampling is some times called chopper sampling because the waveform of the sampled signal appears to be chopped off from the original signal waveform. * Explanation of) UUL Let us consider an analog continuous . time signal xf) to be sampled at the xtt) s(t) rate of f, Hz and f, is higher than Nyquist rate such that sampling theorem is satisfied. A sampled signal Fig. 1.4.2 Natural samplerDigital Communications 1-34 Pulse Digital Modulation s(t) is obtained by multiplication of a sampling function and signal x(t). Sampling function c(t) is a train of periodic pulses of width + and frequency equal to f, Hz. Fig, 1.4.2 shows a functional diagram of natural sampler. When dt) goes high, a switch ‘s' is closed. Therefore, st) = x6) when c(t) = A st) = 0 when c() = 0 Here A is amplitude of c(t). The waveforms of x{'), c(t) and s(t) are shown in Fig. 1.43 (a), 143 (b) and 1.43 () respectively. Signal s(t) can also be defined mathematically as, xt) t (b) ) Fig. 1.4.3 (a) Continuous time signal x(t) (b) Sampling function waveform Le. periodic pulse train {c) Naturally sampled signal waveform s(¢)Digital Communications 4-35 Pulse Digital Modulation sit) = oft) -a(t) oe (1.4.4) Here, c(t) is the periodic train of pulses of width t and frequency f,. Spectrum of Naturally Sampled Signal * Exponential Fourier Series for a periodic waveform is given as att) =} c, ef2nt/To oe For the periodic pulse train of c(t) we have, ht “7 = Period of e(#). ee fo = fp=at=d. = Frequency of c(t). 0 Is ™ T Above equation will be, [with x()= d(], at) = Sc, eltusee o (145) c(t) is a rectangular pulse train. C,, for this waveform is given as : c, = TAsinc(f, 7) To Here T = Pulse width = « and fy = Harmonic frequency. Here f, =f, or C, = Asine (fn) ws (146) T, <.Fourier series for periodic pulse train will be written from equation 145 and equation 1.4.6 as, at) = ¥ Asincys, 0) efit (147) none Js On putting the value of () in equation 1.4.4 we get, s(t) = pa S, sinc fr) e286" -240) o (147 (a) cee This equation represents naturally sampled signals. Now Fourier transform of s(t) is obtained by definition of FT as,Digital Communications 1-36 Pulse Digital Modulation Sf) = FT Ia(b) 4 a yr sine fun) FT {oi2R6% « 0} . (148) We know from frequency shifting property of Fourier transform that, eR at) Oo Xf) a (1.49) A SG Si) = a E sinc(f.t) Xf - fon) (1.4.10) We know that f, =f, ie. harmonic frequency 2 Above equation becomes, Spectrum of Naturally Sampled Signal : $(f)=<" J} sinc (nf,0 XUf - nf,) w (14.11) Comments ; () XY) are periodic in f and are weighed by the sinc function. Fig. 144 (a) shows some arbitrary spectra for x(t) and corresponding spectrum S(f) is shown in Fig. 14.4 (b). (i) Thus unlike the spectrum of instantaneously sampled signal given by Fig.1.3.2 (b), the spectrum of naturally sampled signal is weighted by sinc function, But spectrum of instantaneously sampled signal given by Fig. 1.3.2 (b) remains constant throughout the frequency range. XH) @ (b) + HO — wow En OT Fig. 1.44 (a) Spectrum of continuous time signal x (¢) (b) Spectrum of naturally sampled signalDigital Communications 1-37 Pulse Digital Modulation 1.4.3 Flat Top Sampling or Rectangular Pulse Sampling Basic Principle This is also a practically possible sampling method. Natural sampling is little complex, but it is very easy to get flat top samples. The top of the samples remains constant and equal to instantaneous value of baseband signal x(t) at the start of sampling. The duration of each sample is t and sampling rate is equal to f, = * Tq Generation of flat top samples Fig.1.4.5 (a) shows the functional diagram of sample and hold circuit generating flat top samples and Fig. 1.4.5 (b) shows waveforms. Sampling switch Discharge switch (bo) Fig. 1.4.5 (a) Sample and hold circuit generating flat top sampling (b) Waveforms of flat top sampling Normally the width of the pulse in flat top sampling and natural sampling is increased as far as possible to reduce the transmission bandwidth. Explanation of Flat top Sampled PAM Here we can see from Fig. 1.4.5 (b) that only starting edge of the pulse represents instantaneous value of the baseband signal x(f). The flat top pulse of s(f) is mathematically equivalent to the convolution of instantaneous sample and pulse h (') as shown in Fig. 1.4.6.Digital Communications Pulse Digital Communications = 1-38 “Pulse Digital Modulation Modulation Eh Fig. 1.4.6 Convolution of any function with delta function is equal to that function « That is width of the pulse in s(t) is determined by width of h(f), and sampling instant is determined by delta function. In the waveforms shown in Fig. 1.4.5 (b), the starting edge of pulse represents the point where baseband signal is sampled and width is determined by function h(t), Therefore s(t) will be given as, s() = x O*hO ew (1.4.12) The meaning of this equation is further explained by Fig. 1.4.7. By the replication property of delta function we know that x()*8Q = x) w= (14.18) This is explained in Fig. 1.46 also. The same property is used to obtain flat top samples. «The delta function in equation 1.4.13 is instantaneously sampled signal x; (1), and function h (f) is convolved with x,(). Clearly observe that we are not directly applying equation 1.4.13 here, but we are using it similarly. In equation 1.4.13, 5 (#) is constant amplitude delta function. But in Fig. 1.4.7 (b), %5() is varying amplitude train of impulses. Therefore on convolution of xg()andh(®) we get a pulse whose duration is equal to h(t) only but amplitude is defined by 2 (0. From equation 1.3.1 x5 (#) is given as, x30 = ¥ xin7,)8¢-n7%) was (14.14) none ~. From equation 1.4.12 we can write the convolution as, s) = x O*hODigital Communications 1-39 Pulse Digital Modulation ie., j x5 (w) h(t-u) du j ¥. x(n 7,)8u-nT,)h(t-1) du From equation (1.4.14) DY xT) J 8@—nT,yhe-w du ww (1.4.15) n= From the sifting property of delta function we know that, xit) 7 (b) TT, 0 Ty 21 aT ATS And) he htt) ha) | Fig. 1.4.7 (a) Baseband signal x (t) (b) Instantaneously sampled signal x; (t) (c) Constant pulse width function h(t) (q) Flat top sampled signal s(t) obtained through convolution of h (t)and x; (t)Digital Communications 1-40 Pulse Digital Modulation j FO8U-to) = ftp) (14.16) Using this equation we can write equation 1.4.15 as, s® = > x(nT,)he-nT,) w= (1.4.17) ie + This equation represents value of s(() in terms of sampled value x(n7,) and function h(t 7,) for flat top sampled signal. we also know from equation 1.4.12 that, s) = x307hO By taking Fourier transform of both sides of above equation, Si) = XH ess (14.18) Convolution in time domain is converted to multiplication in frequency domain. Xz () is given as, x= ¥xU-nf) wn (1.4.19) neo Equation 1.4.18 becomes, Spectrum of Flat Top Sampled Signal S()=f, S X(F-nf,) HY) | 1420) naa This equation represents the spectrum of flat top sampled signal. 4.4.3.1 Aperture Effect Definition The spectrum of flat top sampled signal is given by equation 1.4.20 above. This equation shows that the signal s(:) is obtained by passing through a filter having transfer function H(f). The corresponding impulse response h(t), in time domain is shown in Fig. 14:8 (a). This pulse is one pulse of rectangular pulse train shown in Fig. 1.47 (c). Every sample of x(t) is convolved with this pulse. Equation 1.4.20 represents that spectrum of this rectangular pulse is multiplied with that of x5 (1). Fig. 1.4.8 (b) shows the spectrum of one rectangular pulse of ht ('). The spectrum of a rectangular pulse is given as, Hf) = tsinc(f pei Azl w. (14.21)Digital Communications 1-441 Pulse Digital Modulation Fig. 1.4.8 (a) One pulse of rectangular pulse train (b) Spectrum of the pulse of Fig. (a) Thus we can see from Fig. 1.4.8 (b) that by using flat top samples an amplitude distortion is introduced in reconstructed signal x(t) from s(). The high frequency rolloff of H(f) acts like a lowpass filter and attenuates upper portion of message spectrum. These high frequencies of x() are affected. This effect is called aperture effect. Compensation for Aperture Effect As the duration ‘1’ of the pulse increases, aperture effect is more prominent. Therefore during reconstruction an equalizer is required to compensate for this effect. As shown in Fig.14.9, the receiver consists of lowpass reconstruction filter with cutoff frequency slightly higher than the maximum frequency in message signal. The equalizer compensates for the aperture effect. It also compensates for the attenuation by a low-pass reconstruction filter. PAM signal Message signal x) s(t) +noise Fig. 1.4.9 Recovering x(t) From equation 14.21 we know that the sample function h(t) acts like a lowpass filter where Fourier transform is given as, H(f) = tsinc(ft)e~ 7 from equation 1.4.21 ww (14.22)Digital Communications 1-42 This spectrum is plotted in Fig. 14.8. Equalizer used in cascade with the reconstruction filter has the effect of decreasing the inband loss of the reconstruction filter as the frequency increases in such a manner as to compensate for the aperture effect. The transfer function of the equalizer is given by, Ken entta Hog (f) = “AR ws» (1.4.23) Here 't,' is the delay introduced by lowpass filter which is equal to 1/2 Kea tsinc(fe « — x _ Taine) Heq (f) = ws (1.4.24) This is the transfer function of an equalizer. 1.4.4 Comparison of Various Sampling Techniques Various sampling techniques can be compared on the basis of their method, noise interference, spectral properties etc. The following table lists some of the important points of comparison. Sr.| Parameter of | Ideal or instantaneous Natural sampling Flat top sampling No.| comparison ‘sampling 1 Principle of it uses multiplication by It uses chopping It uses sample and sampling an impulss function principle hold circuit 2 | Circuit of sampler # Somaperd, Diocharae re ot) ry ' xt) i t ep toa xq) a(t st] : ieb | 11 4 3 | Waveforms i a ao xl) it) xt) t ee arines an ‘ ‘This is not practically This method is used This method is used possible method practically practicallyDigital Communications 1-43 Pulse Digital Modulation ‘Sampling rate ‘Sampling rate tends to rate satisfies infinity. Nyqui Noise Noise interference is | Noise intertorence is | Noise interference is interference maximum minimum maximum Time domain repre-| oa As = sentaion | %5(t)= 2 s=7 2 sit= 2 x (0T,)8(t-0Ts) x (sinc (nf, 1) oi 2xnfet x (AT, )h ((-ATe) xt, & see so=f = XC =At)HO) X (f=nf,) sinc (1 f,t)X (f nf.) Table 1.4.1 Comparison of sampling techniques ‘> Example 1.4.1 : The spectrum of signal x(t) is shown below. This signal is sampled at the Nyguist rate with a periodic train of rectangular pulses of duration 50/3 milliseconds. Find the spectrum of the sampled signal for frequencies upto 50 Hz giving relevant expression. 3 =i0 0 10 t Fig. 1.4.10 Solution : It is clear from Fig. 1.4.10 that the signal is bandlimited to 10 Hz. W = 10Hz Nyquist rate = 2xW=2x10=20Hz Since the signal is sampled at Nyquist rate, the sampling frequency will be, fe = 20Hz Rectangular pulses are used for sampling. That is flat top sampling’ is used. The spectrum of flat top sampled signal is given by equation 1.4.20 as, sp =h DxG-nAHO (14.25)Digital Communications 1-44 Pulse Digital Modulation Value of H(f) is given by equation 1.4.21 as, H(f) = tsine (fren tft v= (1.4.26) Here t is the width of the rectangular pulse used for sampling, The given value of rectangular sampling pulse is 50/3 milliseconds. ie, = B10 0.05 or tS ag seconds Putting the value of t in equation 1.4.26 we get, Jriossurs Put this value of H(f) and f, in equation 1.4.25 Sif) = 20 E x1 20058 sine{ GY )e-foaers (Since f, =20) si = 5 s x¢p-20mpxsin( OE J005KF/3 ‘This expression gives the spectrum up to 60 Hz (since n=+3) for the sampled signal. ‘=> Example 1.4.2 : A flat top sampling system samples a signal of maximum 1 Hz with 2.5 Hz sampling frequency. The duration of the pulse is 0.2 seconds. Calculate the amplitude distortion due to aperture effect at highest signal frequency. Also find out the equalization characteristic. Solution : It is given that Sampling frequency f= 25 Hz Maximum signal frequency finay = 1 Hz Pulse width t= 02 sec. By equation 1.4.22, the aperture effect is given by a transfer function H (f) as, H(f) = tsinc (frye inftDigital Communications 1-45 Pulse Digital Modulation The magnitude of the above equation is given as, [H(@| = tsinc(f) vo (1.4.27) JH(P| = O2sinc(fx 02) Aperture effect at highest frequency will be obtained by putting f =fiy,, =1Hz in above equation ie., |H@| = 0.2 sine (0.2) = 0.18709 or |H@| = 18.70% w+ (Ans) From equation 1.4.24 the equalizer characteristic is given as, _ _k He = t sinc (ft) Putting t =0.2second and assuming k=1, the above equation will be, = 1 Ha = O2sinc (02 A) a (1.4.28) This equation is the plot of H,,(f)Vsf and it represents the equalization characteristic to overcome aperture effect. 1.4.5 Transmission Bandwidth of PAM Signal The pulse duration ‘1’ is supposed to be very very small compared to time period T, between the two samples. If the maximum frequency in the signal x(t) is 'W' then by sampling theorem, f, should be higher than Nyquist rate or, fe 2 Wor 1. a1 TS apy since f= 7 1 and t << Rsay (1.4.29) If ON and OFF time of the pulse is same, then frequency of the PAM pulse becomes, _ 1 1 =o (14.30) : - Thus Fig. 14.11 shows that if ON and OFF times of PAM signal are Fig. 1.4.11 inmate Sequentey et PAM same, then maximum frequency of signalDigital Communications 1-46 Pulse Digital Modulation PAM signal is given by equation 1.4.30 ie., 1 Jmax = 9% w= (1.4.31) ». Bandwidth required for transmission of PAM signal will be equal to maximum frequency fmax given by above equation. This bandwidth gives adequate pulse resolution i.e., Br 2 fmax Bre w (1.432) Siner> W w= (1433) Thus the transmission bandwidth By of PAM signal is very very large compared to highest frequency in the signal x(0. 1.4.6 Disadvantages of PAM 1. As we have seen just now, the bandwidth needed for transmission of PAM signal is very very large compared to its maximum frequency content. 2. The amplitude of PAM pulses varies according to modulating signal. Therefore interference of noise is maximum for the PAM signal and this noise cannot be removed very easily. 3. Since amplitude of PAM signal varies, this also varies the peak power required by the transmitter with modulating signal. Theory Questions 1. Distinguish between instantaneous sampling, natural sampling and flat top sampling. With functional block diagram explain the working of a circuit that provides flat top sampling. 2. Show that a bandlimited signal of finite energy, which has no frequency components higher than W Hz may be completely recovered from the knowledge of its samples taken at the rate of 2W samples per second. How the recovered signal will differ in amplitude if samples are taken by (a) Natural sampling (b) Flat top sampling ? 3. What is aperture effect ? How it can be reduced ?Digital Communications Pulse Digital Modulation 1.5 Other Forms of Pulse Modulation There are two more types of pulse modulation other than PAM : (i) Pulse Duration Modulation (PDM) In this technique the width of the pulse changes according to amplitude of the modulating signal at sampling instant. Fig. 1.5.1 (c) shows such signal. (ii) Pulse Position Modulation (PPM) In this technique the position of the pulse changes according to amplitude of the modulating signal of sampling instant. Fig. 1.5.1(d) shows such signal. (a) Flat Top PAM (b) (c) PPM. Fig. 1.5.1 Various pulse modulation methods + Pulse position modulation (PPM) and pulse duration modulation (PDM or PWM) both modulate the time parameter of the pulses. PPM has fixed width pulses where as width of PDM pulses varies. Both the methods’ are of constant amplitude.Digital Communications 1-48 Pulse Digital Modulation 1.5.1 Generation of PPM and PDM The block diagram of Fig. 1.5.2 (a) shows the scheme to generate PDM and PPM. The corresponding waveforms are shown in Fig. 1.5.2 (b). The scheme of Fig.1.5.2(a) combines both sampling and modulation operation. The sawtooth generator generates the sawtooth signal of frequency f, (ie. period T,). The sawtooth signal, also called sampling signal is applied to the inverting input of comparator. Comparator xt) © PDMIPWM (a) o PPM (b) Fig. 1.5.2 Generator of PPM and PDM (a) Block diagram (b) Waveforms The modulating signal x(#) is applied to the noninverting input of the comparator. The output of the comparator is high only when instantaneous value of x(f) is higher than that of sawtooth waveform. Thus the leading edge of PDM signal occurs at the fixed time period ie. KT, the trailing edge of output of comparator depends on the amplitude of signal x(t). When sawtooth waveform voltage is greater than voltage of x(® at that instant, the output of comparator remains zero. The trailing edge of the output of comparator (PDM) is modulated by the signal x(f). If the sawtooth waveform is reversed, then trailing edge will be fixed and leading edge will beDigital Communications 1-49 Pulse Digital Modulation modulated. If sawtooth waveform is replaced by triangular waveform, then both leading and trailing edges will be modulated. The pulse duration modulation (PDM) or PWM signal is nothing but output of the comparator. The amplitude of this PDM or PWM signal will be positive saturation of the comparator, which is shown as ‘A' in the waveforms. The amplitude is same for all pulses. To generate pulse position modulation (PPM), PDM signal is used as the trigger input to one monostable multivibrator. The monostable output remains zero untill it is triggered. The monostable is triggered on the falling (trailing) edge of PDM. The output of monostable then switches to positive saturation level ‘A’. This voltage remains high for the fixed period then goes low. The width of the pulse can be determined by monostable. The pulse is this delayed from sampling time KT, depending on the amplitude of signal x(t) at kT;. 1.5.2 Transmission Bandwidth of PPM and PDM ‘As can be seen from the waveform, both PPM and PDM possess DC value. The amplitude of all the pulses is same. Therefore nonlinear amplitude distortion as well as noise interference does not affect the detection at the receiver. However both PPM and PDM needs a sharp rise time and fall time for pulses in order to preserve the message information. Rise time should be very very less than T, ie., << And transmission bandwidth should be, 1 Br > a Thus the transmission bandwidth of PPM and PDM is higher than PAM. The power requirement of PPM is less than that of PDM because of short duration pulses. It can be further reduced by transmitting only edges rather than pulses. Transmission bandwidth of PDM and PPM : By = (151) if 1.5.3 Comparison between Various Pulse Modulation Methods Following table shows the comparison among various pulse modulation techniques.Digital Communications Pulse Digital Modulation Sr. |Pulse Amplitude Modulation No. Pulse Width/Duration Modulation Pulse Position Modulation 4 Wavetorm # % ee Waveform ‘Waveform Time Amplitude of the pulse is proportional to amplitude of modulating signal. Width of the pulse is proportional to amplitude of modulating signal. The relative position of the pulse is proprotional to the amplitude of modulating signal 3] The bandwidth of the transmission channel depends ‘on width of the pulse. Bandwidth of transmission channel depends on rise time of the pulse. Bandwidth of transmission channel depends on rising time of the puise. 4 | The instantaneous power of the transmitter varies. The instantaneous power of the transmitter varies. ‘The instantaneous power of the transrritter remains constant. 5 | Noise interference is high Noise interference is minimum. Noise interference is minimum. 6 | System is complex, Simple to implement. Simple to implement. Simi to frequency modul to phase modul Table 1.5.1 Comparison of PAM, PPM and PDM ww> Example 1.5.1: For a PAM transmission of voice signal with W = 3 kHz. Calculate Br if f, =8kHz and t=01T,. Solution : 1 T, is given as, T, = T 04 T, = 1 t- 1 scc fe 8x105 1 8x105 sec From equation 15.1, the transmission bandwidth Br is given as, 1 By = == 2t 2x o1 1 Lge 8x103Digital Communications 41-51 Pulse Digital Modulation mmm> Example 1.5.2: For the signal given in example 1.5.1, if the rise time is 1% of the width of the pulse, find out the minimum transmission bandwidth needed for PDM and PPM. Ol Solution : In example 1.5.1 we obtained the pulse width t x10? x sec. The rise time is given as 1% of width of pulse ie., = 1x00) = x 901 = 1.251077 sec 8x10° We know that transmission bandwidth is given as, ap & eh = 24 MHz ty © 21.25 x107 Theory Questions 1. Compare PAM, PPM and PDM. 2. Explain the scheme to generate PDM and PPM. 3._Explain how to generate PAM signal for various types of sampling techniques. 1.6 Bandwidth Noise Trade-off The noise analysis of PPM and FM have similar results as follows : i) For both systems, the figure of merit is proportional to square of the ratio Br (i): ii) As the signal to noise ratio is reduced, both the systems exhibit threshold effect. * With digital pulse modulation, the better noise performance than square law can be obtained. The digital pulse modulation such as pulse code modulation gives negligible noise effect by increasing the average power in binary PCM signal. * With PCM, the bandwidth noise trade-off can be related by exponential law.Digital Communications 1-52 Pulse Digital Modulation 1.7 Time Division Multiplexing (PAMITDM System) In PAM, PPM and PDM the pulse is present for short duration and form most of the time between the two pulses, no signal is present. This free space between the pulses can be occupied by pulses from other channels. This is called Time Division Multiplexing (TDM). It makes maximum utilization of the transmission channel. 1.7.1 Block Diagram of PAM / TDM Fig.1.7.1 (a) shows the block diagram of a simple TDM system and Fig. 1.7.1 (b) shows the waveforms of the system. The system shows the time division multiplexing of 'N' PAM channels. Each channel to be transmitted is passed through the lowpass filter. The outputs of the lowpass filters are connected to the rotating sampling switch or commutator. It takes the sample from each channel per revolution and rotates at the rate of f,. Thus the sampling frequency becomes f,. The single signal composed due to multiplexing of input channels is given to the transmission channel. At the receiver the decommutator separates (decodes) the time multiplexed input channels. These channel signals are then passed through lowpass reconstruction filters. Inputs LPFs LPFs ‘Outputs Mattiplexed PAM wave recy tele Fig. 1.7.1 TOM system (PAM/TDM system) (a) Block diagram —_(b) WaveformsDigital Communications 1-53 Pulse Digital Modulation If the highest signal frequency present in all the channels is 'W’, then by sampling theorem the sampling frequency f, should be, fe = 2W w= (L7.) Therefore the time space between successive samples from any one input will be 1 lee wo (1.7.2) ; (1.7.2) a % < oy = (173) Ss Thus the time interval T, contains one sample from each input. This time interval is called frame. Let there be ‘N' input channels. Then in each frame there will be one sample from each of the 'N’ channels. That is one frame of T, seconds contain total 'N’ samples. Therefore pulse to pulse spacing between two samples in the frame will be T, equal to 55. T, .. Spacing between two samples = W w- (1.7.4) nt” channel pulse (nt) channel pulse TN TN Fig. 1.7.2 Calculation of number of pulses per second in TDM From the above figure we can very easily calculate the number of pulses per second or pulse frequency as, 1 Spacing beiween two pulses Number of pulses per second= 1 JN Az 4Digital Communications 1-54 Pulse Digital Modulation We know that I N 1/f :. Number of pulses per second = =Nf « (1.7.5) These number of pulses per second is also called signalling rate of TDM signal and is denc..d by 'r' ie, Signalling rate = r=N (1.7.6) Since fi 2 2W, then signaling rate becomes, ignalling rate in PAM/TDM system w. 17.7) The RF transmission of TDM needs modulation. That is TDM signal should modulate some carrier. Before modulation, the pulsed signal in TDM is converted to baseband signal. That is pulsed TDM signal is converted to smooth modulating waveform x, (; the baseband signal that modulates the carrier. The baseband signal x, (#) passes through all the individual sample values baseband signal is obtained by passing pulsed TDM signal through lowpass filter. The bandwidth of this lowpass filter is given by half of the signalling rate. i.e., _11 By = Zra5NK .. (1.7.8) . Transmission bandwidth of TDM channel will be equal to bandwidth of the lowpass filter, 3N fi from above equation If sampling rate becomes equal to Nyquist rate i.e., f, (min) = Nyquist rate = 2W, then Br = $NxaW Minimum transmission bandwidth of TDM channel : By = NW “G7 This equation shows that if there are total 'N’ channels in TDM which are bandlimited to 'W' Hz, then minimum bandwidth of the transmission channel will be equal to NW.Digital Communicatio! 1-55 Pulse Digital Modulation um Example 1.7.1: 'N’ number of independent baseband signal samples are transmitted over a channel of bandwidth = f. Hz. If each sample is bandlimited to f,, Hz, show that the channel need not have a bandwidth larger than Nf, in order to avoid crosstalk, Solution : Here we have to show that, the bandwidth of the transmission channel in PAM/TDM system should be minimum of Nf, in order to avoid crosstalk between successive channel samples. From Fig. 1.7.1 we know that samples from various channels are interlaced one after another. The figure is reproduced here for convenience. Impuises from various channel a samples % x a ees L2NTa! (one frame) Fig. 1.7.3 PAMITDM samples with instantaneous sampling Here we will assume that the samples from various channels are instantaneously sampled. Thus the samples are impulses of various height. One frame is of 'T,' duration. In this frame there are impulses from ‘N’ channels. Therefore the time space between any two consecutive samples will be, Spacing between two consecutive samples = % wo (17.10) Since maximum signal frequency is f,, the minimum sampling frequency will be f; =2%fm (Le. minimum sampling rate or Nyquist rate). Fret fn Therefore equation 1.7.10 will be, t= 1 2N fn ‘The impulse train of Fig. 1.73 is given to PAM/TDM transmission channel. This channel is lowpass type of channel as shown in Fig. 17.4. Spacing between two consecutive samples = e (17.11)Digital Communications 1-56 Pulse Digital Modulation X x, x % f impulses from various channels Lowpass type transmission channel Fig. 1.7.4 PAM/TDM transmission channel As shown in the above figure, the transmission channel is lowpass type and it has bandwidth of 'f,’ Hz. Therefore it is approximated by an ideal lowpass filter response. The response of the channel is | H(f)|=1 over -f. $f < f.. The input x()) to the transmission channel are impulses from various channels. Those impulses are passed through the transmission channel. Hence output y(t) will be impulse response of the transmission channel. We know that the transfer function H(f)is the Fourier transform of impulse response h (f). Therefore, Impulse response of the transmission channel = h (() = IFT [H (f)] Since output y (¢) is nothing but impulse response of transmission channel (since input x(!) is train of impulses), y¥® = hO=ETIH() = J Heit af By definition of IFT. 4 . = f Lest af Since H (f)=1 for -f. $f $ fe “fe [= i __ etal — e~ Pale Pert | jont _ 1 [eimit — eit =3\— a = a sin (2xf.1) [By Euler's theorem] 1.7.12) = 2f, sin ry) By rearranging the equation nf. 2f,. sinc (2f.t) (17.13)Digital Communications 1-57 Pulse Digital Modulation Thus the output is a sinc function and we know that it has zero values when 2ft = £142,43,24,. 1 z a 4 greta te tees 2 Ofe' 2fe'” fe This can also be verified from equation 17.12. At above given values of t, sin (2xf,t) has zero values, Fig. 1.7.5 shows the plot of sinc function, ie. tek The amplitudes of sinc pulses are weighed by the amplitudes of their impulses. Responses due to various impulses go to zero at these poinis ms OPT ore Impulse Impulse SPacing to avoid due tox, is due to x,is “Foss talk appied here applied here Fig. 1.7.5 Signal at the output of transmission channel which has a bandwidth of f, Hz Thus if impulse from channel X, is applied at f=0, then its corresponding output (ie. its impulse response given by equation 1.7.13) is shown by solid line in above figure. It shows that the response due to one impulse at t= 0 persists over a long time. Consider that second impulse due to second channel is applied at tae The response due to this impulse also persists over long period. This means at time the responses due to other impulses are present. Therefore there is possibility of crosstalk. But a careful obigevation of Fig. 1.7.5 shows that responses due to all the ‘ 2 3 4 other impulses preerpe a. ste ts, P BB Oe BE at that time. For example at {= 0, responses due to all other impulses are zero except . except that of impulse sentDigital Communications 4-58 Pulse Digital Modulation impulse response due to x1, it has peak value of t=0. Similarly at tgp impulse response due to x, is at peak vans all oan responses are zero. This om that if a be ee other words we can say that the spacing between two consecutive samples should be ¥ in order to avoid crosstalk, ie., impulses are transmitted at {=0, += the crosstalk will be zero. In spacing between two consecutive samples in order to avoid crosstalk = <1 2, vn (1.7.14) Comparing the above equation with equation 17.11 (which also gives spacing between two consecutive samples), Jj. 1 2f. 2 fin fo = NSn Thus, Minimum channel bandwidth to avoid crosstalk: f,=N fy vu (1.7.15) Observe that this equation is similar to the relation we obtained earlier given by equation 1.7.9. 1.7.2 Synchronization in TDM System From the discussion of TDM system it is clear that the receiver should operate in perfect synchronization with the transmitter. Normally markers are inserted to indicate the separation between the frames. Fig. 1.7.6 shows the TDM signals with markers. Marker pulse Marker pulse [+ o%0 tame ——+ Fig. 1.7.6 Marker pulses for synchronization in TOM The above figure shows that a marker pulse is inserted at the end of the frame. Because of the marker pulse, synchronization is obtained but number of channels to be multiplexed is reduced by one (ie. N-1 channels can be multiplexed).Digital Communications 1-59 Pulse Digital Modulation 1.7.3 Crosstalk and Guard Times We have seen that RF transmission of TDM needs modulation. Hence the TDM signal is converted to a smooth modulating waveform (i.e. baseband signal) by passing through a baseband filter. Fig. 1.7.7 shows the TDM transmission with baseband filtering and the baseband waveform. x (a XN. Baseband Filter By = HN, Holt) xt) f, carrier %elt) (b) xp, Fig. 1.7.7 (a) TOM transmission with baseband filtering (b) Baseband waveform Thus the baseband waveform passes through the values of all the individual samples. The baseband filtering gives rise to interchannel crosstalk from one sample value to the next. In other words crosstalk means the individual signal sample amplitudes interfere with each other. This interference can be reduced by increasing the distance between individual signal samples. The minimum distance between the individual signal samples to avoid crosstalk is called guard time. Now let us derive an expression for guard time in TDM. Let us assume that the transmission channel acts like a first order lowpass filter with 3-dB bandwidth 'B’. And assume that every pulse transmitted in TDM is a rectangular pulse. When this pulse is applied to the channel, its response is shown in Fig. 1.7.8 (b). In the Fig. 1.7.8 observe that even after the pulse is removed, the response of the channel decays from its value of 'A'. The response then decays for long period. The guard time T, represents the minimum pulse spacing. At the end of guard time, the value of pulse tail is less than A,,,,where it is given as, A ae = Aes + (17.16)Digital Communications 1-60 Pulse Digital Modulation This decay gives rise to crosstalk suactine Fig. 1.7.8 {a} A rectangular pulse applied to the ss channel Response of the lowpass channel to the rectangular pulse And the cross talk reduction factor is defined as, 2 Ky = 10 wast] ~ -54.5BT, dB v (LTA) This equation shows that to keep cross talk below -30dB,T, should be greater than 55. The guard times are very much important particularly in pulse duration or prise pe position modulation techniques. ‘=> Example 1.7.2 : Twelve different message signals, each of bandwidth 10 kHz are to be multiplexed and transmitted. Determine the minimum bandwidth required for PAM/TDM system. Solution : Here the number of channels N = 12. Bandwidth of each channel f,, = 10 kHz Minimum channels bandwidth to avoid crosstalk in PAM/TDM system is, fo = Nn (By equation 1.7.15) 12x 10kHz 120 kHz mm> Example 1.7.3: Twenty four voice signals are sampled uniformly and then time division multiplexed. The highest frequency component for each voice signal is 3.4 kHz.Digital Communications 1-61 Pulse Digital Modulation 1) If the signals are pulse amplitude modulated using Nyquist rate sampling, what is the minimum channel bandwidth required? ii) If the signals are pulse code modulated with an 8 bit encoder, what is the sampling rate ? The bit rate of system is 1.5x10° bits/sec. Solution : i) We know that if N channels are time division multiplexed, then minimum transmission bandwidth is given as, Br = NW Here W is the maximum frequency in the signals. By = 24x34 kHz=816kHz ws (Ans) ii) The signalling rate of the system is given as, r = 1.5x10® bits/sec Since there are 24 channels, the bit rate of an individual channel is, 1.5106 (one channel) = = 62500 bits/sec Since each sample is encoded using 8 bits, the samples per second will be, __ F (one channel) bits / sec Sample/eec = "Ti yaampie) Samples per seconds is nothing but sampling frequency. 62500 bits/ sec he & > Sbits/sample = 7812.5 Hz or samples per second ws. (Ans) map Example 1.7.4: Twenty four voice signals are sampled uniformly and then time division multiplexed. The sampling operation uses flat samples with 1 yisec duration. The multiplexing operation provides for synchronization by adding an extra pulse of 1jisec duration. Assuming sampling rate of 8 kHz, calculate spacing between successive pulses of multiplexed signal and setup a scheme for accomplishing a multiplexing requirement. Solution : There are 24 voice signal pulses plus one synchronization pulse. Hence there are total 25 pulses. Sampling rate is 8 kHz. Hence duration of one frame will be, ~ili1 Ts = = 3000 125 jisecDigital Communications 1-62 Pulse Digital Modulation Thus in 125 psec time there are 25 pulses at uniform distances. This is illustrated in Fig. 1.7. | Ts = 125 ms, one relation of sampling swich po i Fig. 1.7.9 Multiplexing of 24 voice signals As shown in above figure, the pulses are separated by “27H = 5 ys. Width of the pulse is 1 ps. Hence, Spacing between pulses = 5 - 1 = 4 psec. Fig. 1.7.10 shows the multiplexing scheme. xe x peace . Multiplexer Mutsenes a7 sampler 200,000 samples er Second %24 Synchronization pulse f,= BkHz 11 usec = Fig. 1.7.10 PAM-TDM system Theory Questions . Explain PAM/TDM system for “N’ number of channels. . Derive the relation for minimum bandwidth to transmit ‘N' channels in PAM/TDM system such that crosstalk is avoided. . Explain the importance of synchronization in TDM systems.Digital Communications 1-63 Pulse Digital Modulation Unsolved Examples 1. Twenty four voice signals are sampled uniformly and then time division multiplexed, the sampling operation uses flat top samples with 1 sec duration. The synchronization is provided by adding an extra pulse of 1 wsec duration. The highest frequency component of each voice signal is 3.4 kHz. (a) For sampling rate of 8 kHz, calculate spacing between successive pulses of multiplexed signal. (2) For Nyquist rate repeat part (a). 1.8 Pulse Code Modulation 1.8.1 PCM Generator The pulse code modulator technique samples the input signal x(f) at frequency >2W. This sampled ‘Variable amplitude’ pulse is then digitized by the analog to ital converter. The parallel bits obtained are converted to a serial bit stream. Fig.1.8.1 shows the PCM generator. Se v digits 220 Fig. 1.8.1 PCM generator In the PCM generator of above figure, the signal x(\) is first passed through the lowpass filter of cutoff frequency 'W' Hz. This lowpass filter blocks all the frequency components above 'W' Hz. Thus x(t) is bandlimited to 'W' Hz. The sample and hold circuit then samples this signal at the rate of f.. Sampling frequency f, is selected sufficiently above Nyquist rate to avoid aliasing i.e., f2w In Fig. 1.8.1 output of sample and hold is called x(nT,). This x(nT,) is discrete in time and continuous in amplitude. A q-level quantizer compares input x(n T,) with its fixed digital levels. It assigns any one of the digital level to x(n 7,) with its fixed digital levels. It then assigns any one of the digital level to x(nT,) which results in minimum distortion or error. This error is called quantization error. Thus output of quantizer is a digital level called x, (1 T,).Digital Communications 1-64 Pulse Digital Modulation Now coming back to our discussion of PCM generation, the quantized signal level x,(nT,) is given to binary encoder. This encoder converts input signal to 'v’ digits binary word. Thus x, (T,) is converted to 'V' binary bits. The encoder is also called digitizer. Tt is not possible to transmit each bit of the binary word separately on transmission line. Therefore ‘0’ binary digits are converted to serial bit stream to generate single baseband signal. In a parallel to serial converter, normally a shift register does this job. The output of PCM generator is thus a single baseband signal of binary bits. ‘An oscillator generates the clocks for sample and hold an parallel to serial converter. In the pulse code modulation generator discussed above ; sample and hold, quantizer and encoder combinely form an analog to digital converter. 1.8.2 Transmission Bandwidth in PCM Let the quantizer use ‘v’ number of binary digits to represent each level. Then the number of levels that can be represented by ‘v’ digits will be, q= 2 = (181) Here ‘q' represents total number of digital levels of q-level quantizer. For example if v= 3 bits, then total number of levels will be, q = 23 =8 levels Each sample is converted to 'v' binary bits. i.e. Number of bits per sample = 0 We know that, Number of samples per second = f, . Number of bits per second is given by, (Number of bits per second) = (Number of bits per samples) x (Number of samples per second) = v bits per sample x f, samples per second. (1.8.2) The number of bits per second is also called signaling rate of PCM and is denoted by ie, Signaling rate in PCM: r = 0f, wo» (183) Here f, 2 2W.Digital Communications 1-65 Pulse Digital Modulation Bandwidth needed for PCM transmission will be given by half of the signaling rate ie., Bre ; r (8.4) Transmission Bandwidth of PCM : Br2 tof Since f, 22W 2+ (18.5) Bp20W (1.8.6) 1.8.3 PCM Receiver Fig. 1.8.2 (a) shows the block diagram of PCM receiver and Fig. 1.8.2 (b) shows the reconstructed signal. The regenerator at the start of PCM receiver reshapes the pulses and removes the noise. This signal is then converted to parallel digital words for each sample. v digits PCM+ Noise ‘Serial FET Digital to parallel toanalog converter} converter () (bo) Fig. 1.8.2 (a) PCM receiver (b) Reconstructed waveformDigital Communications 1-66 Pulse Digital Modulation The digital word is converted to its analog value x, () along with sample and hold. This signal, at the output of S/H is passed through lowpass reconstruction filter to get yp (). As shown in reconstructed signal of Fig. 1.82 (b), it is impossible to Teconstruct exact original signal x(!) because of permanent quantization error introduced during quantization at the transmitter. This quantization error can be reduced by increasing the binary levels. This is equivalent to increasing binary digits (bits) per sample. But increasing bils 'v' increases the signaling rate as well as transmission bandwidth as we have seen in equation 1.8.3 and equation 1.8.6. Therefore the choice of these parameters is made, such that noise due to quantization error (called as quantization noise) is in tolerable limits. 1.8.4 Uniform Quantization (Linear Quantization) We know that input sample value is quantized to nearest digital level. This quantization can be uniform or nonuniform. In uniform quantization, the quantization step or difference between two quantization levels remains constant over the complete amplitude range. Depending upon the transfer characteristic there are three types of uniform or linear quantizers as discussed next. 1.8.4.1 Midtread Quantizer ‘The transfer characteristic of the midtread quantizer is shown in Fig. 183. As shown in this figure, when an input is between - 8/2 and + 5/2 then the quantizer output is zero. i.e., For -8/2 < x(nT) < 3/2; x, (nT) = 0 Here ‘& is the step size of the quantizer. for 8/2 < x (al) <38/2; xy (nT) =8 Similarly other levels are assigned. It is called midtread because quantizer output is zero when x(nT,) is zero. Fig.1.8.3 (b) shows the quantization error of midtread quantizer. Quantization error is given as, € = xq (nT,) - x (aT) (18.7) In Fig. 1.8.3 (b) observe that when x(nT,) = 0, x,(n\T,) = 0. Hence quantization error is zero at origin. When x(nT,) = 8/2, quantizer output is zero just before this level. Hence error is 8/2 near this level. From Fig. 1.8.3 (b) it is clear that, -8/2 < €<8/2 -- (1.8.8) Thus quantization error lies between ~ 6/2 and + 6/2. And maximum quantization -(3} w= (18.9) error is, maximum quantization error, €max =|5Digital Communications 1-67 Pulse Digital Modulation transfer characteristic passes through zero — { 1 t 1 Staircase approximation i ‘ | | Input x{0T) | 58/2 | 782, i Input xiaT,) — Fig. 1.8.3 (a) Quantization characteristic of midtread quantizer (b) Quantization error 1.8.4.2 Midriser Quantizer The transfer characteristic of the midriser quantizer is shown in Fig. 1.8.4. In Fig. 1.84 observe that, when an input is between 0 and 8, the output is 8/2. Similarly when an input is between 0 and ~ 6, the output is - 8/2. ie, For 0 < x (nT) <8; xq (nT) = 8/2 -8S x (AT) <0; x (nT) = -8/2 Similarly when an input is between 38 and 4 8, the output is 7 8/2. This is called midriser quantizer because its output is either + 8/2 or ~ 8/2 when input is zeroDigital Communications Pulse Digital Modulation Fig. 1.8.4 (a) Transfer characteristic of midriser quantizer {b) Quantization error Fig. 1.84 (b) shows the quantization error in midriser quantization. When input x(nT,) = 0, the quantizer will assign the level of 5/2. Hence quantization error will be, € = xq (nT,) - x (nT) = 8/2-0=8/2 Thus the quantization error lies between - 8/2 and + 8/2. ie, -8/2 < es6/2 - (1.8.10) And the maximum quantization error is, Emax = \3| sw» (1.8.11)Digital Communications 1-69 Pulse Digital Modulation In both the midriser and midtread quantizers, the dotted line of unity slope pass through origin. It represents ideal nonquantized input output characteristic. The staircase characteristic is an approximation of this line. The difference between the staircase and unity slope line represents the quantization error. 1.8.4.3 Biased Quantizer Fig. 1.8.5 shows the transfer characteristic of biased uniform quantizer. TTT I | Quantizer output PTT TTT TTT rp a(0Ts) I 1 | + | rhea Cel | 1-| oh Input x(nT,) Fig. 1.8.5 (a) Biased quantizer transfer characteristic (b) Quantization error The midriser and midtread quantizers are rounding quantizers. But biased quantizer is truncation quantizer. This is clear from above diagram. When input is between 0 and 6, the output is zero. ie, for 0 < x(aT)<8; xq (nT) =0Digital Communications 1-70 Pulse Digital Modulation Similarly, for -8 < x (nT) <0; xq (nT) =-5 Fig. 1.85 shows quantization error. When input is 6, output is zero. Hence quantization error is, € = xq (al) - x(aT,) = 0-8=-6 Thus the quantization error lies between 0 and - 8. ie, -8< <0 (1.8.12) And the maximum quantization error is, Emax = | 8] ww (1.8.13) Thus the quantization error is more in biased quantizer compared to midriser and midtread quantizers. The unity slope dotted line passes through origin. It represents ideal nonquantized transfer characteristic. The difference between staircase and dotted line gives quantization error. 1.8.5 Quantization Noise and Signal to Noise Ratio in PCM 4.8.5.1 Derivation of Quantization Error/Noise or Noise Power for Uniform (Linear) Quantization Step 1: Quantization Error Because of quantization, inherent errors are introduced in the signal. This error is called quantization error. We have defined quantization error as, € = x, (nT.)-x(nT,) 1.8.14) Step 2: Step size Let an input x(7 7) be of continuous amplitude in the range —xmax (0 +Xmax- From Fig. 1.84 (a) we know that the total excursion of input x(mT.) is mapped into ‘levels on vertical axis. That is when input is 46, output is 48 and when input is -48, output is -38, That is +xmaq represents Fb and-xpax represents ~76. Therefore the total amplitude range becomes, Total amplitude range = max ~ (- Xmax) Dx page 1.8.15) If this amplitude range is divided into ‘g levels of quantizer, then the step size ‘8 is given as, a) .. (1.8.16)Digital Communications 1-71 Pulse Digital Modulation If signal x(f) is normalized to minimum and maximum values equal to 1, then Xmax = 1 Xmax = -1 w= (1.8.17) Therefore step size will be, 8= : (for normalized signal) .. (1.8.18) Step 3 : Pdf of Quantization error If step size ‘8 is sufficiently small, then it is reasonable to assume that the quantization error ‘e' will be uniformly distributed random variable. The maximum quantization error is given by equation 18.11 as, Emax = Bi +» (1.8.19) ie. -3 = Emax 23 a» (1.8.20) Thus over the interval (-2.4) quantization error is uniformly distributed random variable £400, fa) Fig. 1.8.6 (a) Uniform distribution (b) Uniform distribution for quantization error In above figure, a random variable is said to be uniformly distributed over an interval (a,b). Then PDF of 'X’ is given by, (from equation of Uniform PDF).Digital Communications 1-72 Pulse Digital Modulation 0 for xSa 1 hee 3 for acxsb 0 for x>b o» (18.21) Thus with the help of above equation we can define the probability density function for quantization error ‘e' as, 0 for es 1 3, <8 ge = lt fr ~hees8 ° for o8 va (1.8.22) Stop 4 : Noise Power From Fig. 1.84 (b) we can see that quantization error ‘e’ has zero average value. That is mean 'm,' of the quantization error is zero. The signal to quantization noise ratio of the quantizer is defined as, S$ __ Signal power (normalized) N ~ Noise power (normalized) ~ (18.23) If type of signal at input i.e, x(f) is known, then it is possible to calculate signal power. The noise power is given as, y2 Noise power = a a» (1.8.24) Here V2... is the mean square value of noise voltage. Since noise is defined by noise random variable ‘e' and PDF f, (€), its mean square value is given as, mean square value = E{e2] = &? =» (18.25) ‘The mean square value of a random variable 'X’ is given as, E(X2]= J x? fx @)dx By definition -» (1.8.26) X2 " Here Ele2] = fer de -- 1.8.27)Digital Communications 1-73 Pulse Digital Modulation From equation 1.8.22 we can write above equation as, 8/2 1 Efe?] = J e? xide 3 -8/2 yey" Fe B13} 4. S| 3 3 = (18.28) <. From equation 1.8.25, the mean square value of noise voltage is, 2 v2, = mean square value = 5 When load resistance, R=1 ohm, then the noise power is normalized i.e, v2 Noise power (normalized) = a [with R =1 in equation 1.8.24] _ 8 /12_ 8 ST Te Thus we have, Normalized noise power 2 or Quantization noise power = 5 ; For linear quantization. or Quantization error (in terms of power) = (1.8.29) 1.8.5.2 Derivation of Maximum Signal to Quantization Noise Ratio for Linear Quantization From equation 1.8.23 signal to quantization noise ratio is given as, S$ _ Normalized signal power N ~ ‘Normalized noise power _ Normalized signal power _. (1830) @? /12) The number of bits 'v' and quantization levels ‘q' are related as, qe? ~- (18.31)Digital Communications 1-74 Pulse Digital Modulation Putting this value in equation 1.8.16 we have, 3 = 2imx (18.32) 7 Pulting this value in equation 1.8.30 we get, S _ Normalized signal power 5 =, Normalised’ signal power 2 2xmox Y 412 7 Let normalized signal power be denoted as 'P’. Se Pg SP N 4 Xfax pe This is the required relation for maximum signal to quantization noise ratio. Thus, Maximum signal to quantization noise ratio : 3-2F. 2 x w= (1.8.33) This equation shows that signal to noise power ratio of quantizer increases exponentially with increasing bits per sample. If we assume that input x(f) is normalized, ie., Reae = 2 + (1.8.34) Then signal to quantization noise ratio will be, x = 3x22xP ww (1.8.35) If the destination signal power 'P’ is normalized, ie., Psi (1.8.36) Then the signal to noise ratio is given as, 2 < 3x22 e» (18.37) Since Xmay=landP<1, the signal to noise ratio given by above equation is normalized. Expressing the signal to noise ratio in decibels, $s s Fi F [5]! = 1006 (§ |r snc poner as. a 10 log yp [3x 2?” S (48+ 6v)dBDigital Communications 1-75 Pulse Digital Modulation Thus, Signal to Quantization noise ratio for normalized values of power : (w jaws as +60) dB 'P' and amplitude of input x(t) o» (18.38) ‘mp Example 1.8.1: Derive the expression for signal to quantization noise ratio for PCM system that employs linear quantization technique. Assume that input to the PCM system is a sinusoidal signal. OR A PCM system uses a uniform quantizer followed by av bit encoder. Show that rms signal to quantization noise ratio is approximately given by (1.8 + 6v) dB. Solution : Assume that the modulating signal be a sinusoidal voltage, having peak amplitude A,,. Let this signal cover the complete excursion of representation levels. The power of this signal will be, P= ye Here V = rms value = [An 2p o- 1.8.39) When R =1, the power P is normalized, ie, Normalized power : Ps 4h with R =1 in above equation. :. Signal to quantization noise ratio is given by equation 1.8.33 as, Ss = 3P. xD N *inax 2 Here P= 4a and Xmax Putting these values in the above equation, S 3x22 21.5x2”Digital Communications 1-76 Pulse Digital Modulation Expressing signal to noise power ratio in dB, $s Ss (5) ts 101080 (5 J 10810 (1.522) = 110g jo (1.5) + 10 log 4p 2” 1.76+20%10x 03 Thus, 5 \ini s Saat (5 \s in PCM: (F js = 1.8 +60 ; for sinusoidal signal (1.8.40) mm Example 1.8.2: A Television signal with a bandwidth of 4.2 MHz is transmitted using binary PCM. The number of quantization levels is 512. Calculate, i) Code word length _ ii) Transmission bandwidth iii) Final bit rate iv) Output signal to quantization noise ratio. {March-2003, 10 Marks] Solution : The bandwidth is 4.2 MHz, means highest frequency component will have frequency of 42 MHz ie,, W = 42 MHz Quantization levels q = 512 i) Number of bits and quantization levels are related in binary PCM as, qe ie. 512 = 2” log 512 = vlog 2 on » = logs2 Tog 2 = 9 bits vw (Ans) Thus the code word length is 9 bits. ii) From equation 1.8.6 the transmission channel bandwidth is given as, Br > oW = 9x42 106 Hz By 2 37.8 MHz vw (Ans) iii) The final bit rate will equal to signaling rate. From equation 1.8.3 signaling rate is given as, r= vf,Digital Communications 1-77 Pulse Digital Modulation Sampling frequency f, 2 2W by sampling theorem. f, 2 2x42MHz since W = 4.2 MHz f 2 84 MHz Putting this value of ',' in equation for signaling rate, r = 9x84x108 = 756x10° bits/sec «» (Ans) From equation 1.8.4 transmission bandwidth is also obtained as, ty 2 Br > }x756%105 _ bits/sec or By 2 37.8 MHz which is same as the value obtained earlier. iv) The signal to noise ratio (e je < 48460 dB < 48+6x9 S 58.8 dB + (Ans) ‘a> Example 1.8.3 : The bandwidth of signal input to the PCM is restricted to 4 KHz. The input varies from -38 V to + 38 V and has the average power of 30 mW. The required signal to noise ratio is 20 dB. The modulator produces binary output. Assume uniform quantization. 4) Calculate the number of bits required per sample ii) Outputs of 30 such PCM coders are time multiplexed. What is the minimum required transmission bandwidth for the multiplexed signal ? Solution : The given value of signal to noise ratio is 20 dB. ‘ s Ss ie. ag = 10reg oy J= 2048 = 100 zeDigital Communications 1-78 Pulse Digital Modulation i) The signal to quantization noise ratio is given as, 2 2 = SE22 By equation 1.8.33 ¥ Here Xmax = 38V, P= 30mW and 3x30x 10-3 22 (38)? 6.98. bits = 7 bits as (Ans) 100 = 2 W ii) The maximum frequency is, W = 4 kHz The transmission bandwidth is given by equation 1.8.6 as, Br = oW Since there are 30 PCM coders which are time multiplexed, the transmission bandwidth will be, By = 30x0-W 2 30x7 x4 kHz = 840 KHz vs (Ans) Signaling rate is two times the transmission bandwidth as given by equation 1.84 ie, Signaling rate r = 840x2 bits/sec = 1680 bits/sec. um Example 1.8.4: The information in an analog signal voltage waveform is to be transmitted over a PCM system with an accuracy of +01% (full scale). The analog voltage waveform has a bandwidth of 100 Hz and an amplitude range of -10 to +10 volts, a) Determine the maximum sampling rate required. b) Determine the number of bits in each PCM word. ¢) Determine minimum bit rate required in the PCM signal. d) Determine the minimum absolute channel bandwidth required for the transmission of the PCM signal.Digital Communications 1-79 Pulse Digital Modulation Solution : Here an accuracy is given as + 0.1%. That is quantization error should be £01%. or the maximum quantization error should be £01% or €. = $01% =+0001 The maximum quantization error for an uniform quantizer is given as, or That is Step size 8 = 2x0.001 = 0.002 The step size, number of levels and maximum value of the signal are related as (By equation 1.8.16) b= 2s Here |ximax| = 10 volts ©. Putting values of 6 and Xpayr 0.002 = 2X10 9 oi = 20. 1 yo02 = 10,000 That is the number of levels are 10,000. a) The maximum frequency in the signal is 100 Hz ie. W = 100 Hz By sampling theorem,minimum sampling frequency should be, f 2 Ww > 2x1002200 Hz s- (Ans) b) We know that minimum 10,000 levels should be used to quantize the signal. If binary PCM is used, then number of bits for each samples can be calculated as, q= 2 Here, q = number of levelsDigital Communications 1-80 Pulse Digital Modulation = bits in PCM, 10,000 = 2” log 19 10,000 = vlog yo 2 or 2 or v = 14 bits v (Ans) ©) From equation 1.8.3 the bit rate or signaling rate is given as, r= vf = 14x200 2 2800 bits per second. d) The transmission channel for PCM is given by equation 1.8.4 as, 1 Bp 2 Gr 1 2 3% 2800 2 1400 Hz +» (Ans) tum Example 1.8.5: A PCM sysiem uses a uniform quantizer followed by a 7-bit binary encoder. The bit rate of the system is equal to 50x10° bits/sec. a) What is the maximum message bandwidth for which the system operates satisfactorily ? b) Determine the output signal to quantization noise ratio when a full load sinusoidal modulating wave of frequency 1 MHz is applied to the input. Solution: a) Let us assume that the message bandwidth be W Hz. Therefore sampling frequency should be, fk = 2W The number of bits v = 7 bits From equation 1.8.3 the signaling rate is given as, re of r 2 7x2W 50x106 = 14W (putting value of r) W < 3.57 MHz =» (Ans)aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.1-83 Pulse Digital Modulation Solution : The signal is uniformly distributed in the range +xm,,. Therefore we can write its PDF (using the Standard Uniform Distribution) as, fx) = 0 for x<~Xmax 1 7 for —Xmax < X Xmax Fig. 1.8.7 shows this PDF, 1) max %max Fig. 1.8.7 PDF of a uniformly distributed random variable The mean square value of a random variable X is given as, X2 = J 2 fear ‘Therefore mean square value of x(t) will be, 1 - 2 J P Way " dx ~*max, - fs op * Brox | 3 Xmax = thx i 3 it een eo The signal power P = “0? [since R =1] 2 Normalized signal power P = =Digital Communications 1-84 Pulse Digital Modulation hax 3 . 2tmae . Step size b= a By equation 1.8.16 34 Xmax Zz (84 Normalized signal power, P = 2 Normalized noise power = e By equation 1.8.29 S _ Normalized signal power «Signal to nois ti = js noise power ra 'N ~ Normalized noise power _ gt /e R712 Since q=2*, above equation will be, S _ 2% f2 § oe (w}8 = 10 ogy 22) dB = 60 This is the required expression for maximum value of signal to noise ratio. ump Example 1.8.9: Consider an audio signal comprised of the sinusoidal term 5 (9) =3.cos (500n#) i) Find the signal to quantization noise ratio when this is quantized using 10 bit PCM. i) How many bits of quantization are needed to achieve a signal to quantization noise ratio of atleast 40 dB ? Solution : Here s(f) = 3.cos (500 nt) That is sinusoidal signal applied to the quantizer. i) Let us assume that peak value of cosine wave defined by s(#) covers the complete range of quantizer. ie. Am = 3V covers complete range.Digital Communications 1-85 Pulse Digital Modulation We know that signal to noise ratio for sinusoidal signal is given by s (x = 18+60 Here 10 bit PCM is used ie., v= 10 (5) = 18+6x10 = 61.8 dB ii) For sinusoidal signal again we will use the same relation. i.e. ie. (we = 18+60dB To get signal to noise ratio of at least 40 dB we can write above equation as, 18+6v > 40 dB v 2 6.36 bits = 7 bits Thus at least 7 bits are required to get signal to noise ratio of 40 dB. im Example 1.8.10: A 7 bit PCM system employing uniform quantization has an overall signaling rate of 56 k bits per second. Calculate the signal to quantization noise ratio that would result when its input is a sine wave with peak to peak amplitude equal to 5. Calculate the dynamic range for the sine wave inputs in order that the signal to quantization noise ratio may be less than 30 dBs. What is the theoretical maximum frequency that this system can handle ? Solution: The number of bits in the PCM system are v = 7 bits Assume that 5 V peak to peak voltage utilizes complete range of quantizer. Then we can find the signal to quantization noise ratio as, (3) = 18+60dB =18+6x7 N = 43.8 dB By equation 1.8.3 signaling rate is given as, r=ofDigital Communications 4-86 Pulse Digital Modulation Putting r =56x 103 bits/second and v =7 bits in above equation we get, 536x103 = 7+ f, Sampling frequency, f, = 8x10) Hz By sampling theorem, f, = 2W Maximum frequency that can be handled is given as, fi . 8000 Wi 8 eg W < 4000 Hz (Ans) ma Example 1.8.11 : The bandwidth of TV video plus audio signal is 4.5MHz. If the signal is converted to PCM bit stream with 1024 quantization levels, determine the number of bits/sec generated by the PCM system. Assume that the signal is sampled at the rate of 20% above nyquist rate. If above linear PCM system is converted to companded PCM, will the output bit rate change? Justify. Solution : The given data is, W = 45 MHz q = 1024 levels The Nyquist rate is, Nyquist rate = 2W =2 x 4.5 =9 MHz The sampling rate is 20% above the nyquist rate. i.e. Sampling rate, f = 1.2 x 9 = 108 MHz We know that quantization levels q and number of bits v are related as, q=2 1024 = 2 v = 10bits The number of bits/sec generated by PCM system is called bit rate or signaling rate. ie., Signaling rate,r = vf, = 10 x 10.8 x 108 bits/sec. 108 x10® bits / sec. Waa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.Digital Communications 1-91 Pulse Digital Modulation shown in Fig. 1.89. That is nonlinear transfer characteristic means compression and expansion curves. Compression Expansion Linear characteristics Expansion at receiver Compression at transmitter Fig. 1.8.9 Companding curves for PCM 1.8.6.4 1 - Law Companding for Speech Signals Normally for speech and music signals a - law compression is used. This compression is defined by the following equation, Zs) = (sen MOD jxl<1 w. (1.8.52) Fig. 1.8.10 shows the variation of signal to noise ratio with respect to signal level without companding and with companding. With compandin, Without companding -40 -30 -20 -10 0 Signal tevel dB —> 10 PCM performance with 1. - law compandingaa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.Digital Communications 1-95 Pulse Digital Modulation The desired signal to noise ratio is 42 dB. Hence rise ae )sti is 42 - 30 = 12 dB We know that (Jt increases by 6 dB for 1 bit. Hence 2 bits are required to increase signal to noise ratio by 12 dB. Hence, p = 10 +2= 12 bits are required Gi) To obtain fractional increase in bandwidth Bandwidth in PCM is given as, 1 Br = $4, Br (10 bits) = $x10xf, 5f, and Bp (12 bits) = Fras f *. Fractional increase in By = § x 100% = 20 % mm Example 1.8.15 : A telephone signal with cutoff frequency of 4 kHz is digitized into 8 bit PCM, sampled at Nyquist rate. Calculate baseband transmission bandwidth and quantization < ratio Solution : Given data is, W = 4kHz v = 8B bits From equation 1.8.6 transmission bandwidth is given as, By = 0W = 4kx8= 32 Kz Telephone signal is nonsinusoidal signal. Its signal to quantization noise ratio is given by equation 1.8.38 as, S yy 7 e+e 4.8 + 6x8 = 52.8 dB.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.Digitai Communications 1-99 Pulse Digital Modulation Analog input Multiplexed PCM data Line. waveform generator Channel Analog PEM, output Puls regenerator N Fig. 1.9.2 TDM/PCM system 1.9.3.2 Multiple Channel Frame Alignment For TDM | PCM (T, System) The multiple channel alignment is very important in TDM/PCM system. Fig. 1.93 shows the TDM frame format of most widely used T1 system. 1sms sane cH CEPT EEEEE nn syne bit ras yup | yO yon 8 SEEDED FEEEEDE CELEB EPID Ten ts rw3aigKs Fig. 1.9.3 Multiple channel frame alignment in Ti systemsaa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.Digital Communications 1-103 Pulse Digital Modulation 24 1, 1.5Mbis voice Channey telephone || Bank channels 1 ; M12 Te F 6.3 Mb/s Digital 1, data fy Mux | 2 channels i wea 1, Z Visual Z telephone ECM Ts ™ pom | 73 channel Fig. 1.9.4 Digital multiplexing of voice telephone channels, digital data, TV etc. for AT & T standard Theory Questions 1. Which are the types of digital multiplexers? 2. Explain the frame structure of T1 system in detail 3. With the help of block diagram explain PCM/TDM system. 1.10 Virtues, Limitation and Modifications of PCM Advantages of PCM (i) Effect of channel noise and interference is reduced. (i) PCM permits regeneration of pulses along the transmission path. This reduces noise interference. (ii) The bandwidth and signal to noise ratio are related by exponential law. (iv) Multiplexing of various PCM signals is easily possible. (v) Encryption or decryption can be easily incorporated for security purpose. Limitations of PCM (i) PCM systems are complex compared to analog pulse modulation methods. (ii) The channel bandwidth is also increased because of digital coding of analog pulses.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.Delta Modulation | | | We have seen in PCM that, it transmits all the bits which are used to code the sample. Hence signaling rate and transmission channel bandwidth are large in PCM To overcome this problem Delta Modulation is used. 2.1 Delta Modulation 2.1.1 Operating Principle of DM Delta modulation transmits only one bit per sample. That is the present sample value is compared with the previous sample value and the indication,whether the amplitude is increased or decreased is sent. Input signal x(f) is approximated to step signal by the delta modulator. This step size is fixed. The difference between the input signal x(!) and staircase approximated signal confined to two levels, ie +8and—6, If the difference is positive, then approximated signal is increased by one step ie. °8. If the difference is negative, then approximated signal is reduced by "8. When the step is reduced, ‘0’ is transmitted and if the step is increased, ‘1’ is transmitted. Thus for each sample, only one binary bit is transmitted. Fig. 2.1.1 shows the analog signal x(!) and its staircase approximated signal by the delta modulator. m0) aa + ee | T LI I = Step size = : ‘cH ima eT, Sampling | period | | CS La | | time dela oti of1ro jhe | Binary one |__| | + bitsequencet = 0] 7/4] 1, i] 7/4/60 Fig. 2.1.1 Delta modulation waveform (2-1)aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.Digital Communications Delta Modulation signal is changed by large amount (6) because of large step size. Fig. 2.2.1 shows that when the input signal is almost flat, the staircase signal u(t) keeps on oscillating by +6 around the signal. The error between the input and approximated signal is called granular noise, The solution to this problem is to make step size small. Thus large step size is required to accommodate wide dynamic range of the input signal (to reduce slope overload distortion) and small steps are required to reduce granular noise. Adaptive delta modulation is the modification to overcome these errors, ym Example 2.2.1: Using predictability theory, prove that transmission of encoded error signal (rather than encoded signal itself is sufficient for reasonable reconstruction of signal. With the help of block schematic suggest any one technique to transmit and receive encoded errors. What are the limitations and advantages of such techniques with reference to linear or uniform PCM ? Solution : Here the technique that uses predictibility theory is basically delta modulation. The output of the accumulator in DM transmitter is given by equation 2.15 as, uenT,) = ul(n—T.| +007.) ~ (222) Here WnT.) = +8 or dsgn[dnT,)] Thus b(rT,) basically represents error signal. Sign of step size '' depends upon whether e(nT,) is positive or negative. Now we will show that the signal can be reconstructed only with the help of encoded error signal, ie, b(nT,) The accumulator of Fig. 2.1.2(b) acts as a delta modulation receiver. u(nT,) is the output of accumulator. For simplicity let us drop 7, in equation 2.2.1 Then we get, u(n) = u(n-1)+(n) w= (2.2.2) Observe that this is recursive equation. Hence u(n ~1) can be calculated as, u(n-1) = u(t -2)+0(n-1) w= (22.3) Hence equation 22.2 becomes, u(n) = wWn-2)+Wn-1)+h(n) a» (22.4) From equation 2.2.3 we can calculate u(7 — 2) as, u(n 2) = u(n-3)+K(n-2) Hence equation 22.4 becomes, un) = u(n=3)+H(n-2)+b(n-1) +n)aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.2-9 Delta Modulation «Slope overload will not occur if, 6 © ef o 2.3.4) in ‘The maximum frequency in the signal is, W = 3KHz Nyquist rate = 2W=2x3kHz = 6 kHz Sampling frequency= 5 times Nyquist rate f, = 5x6kHz = 30 kHz onc i 1 1 Sampling interval T, = + =-—~~> oe Js 30x10? Step size 8 = 250mV=250x 109 = 025V Given that f,, = 2kHz=2x109 Hz «Putting these values in equation 2.3.4. 025 2nx 2x103 x A, 30x 103 An $ 0.6 volts ‘mp Example 2.3.3 : With reference to delta modulation System shown in Fig. 2.3.3 show that the optimum step size 22A Kp = RT Sn where A is amplitude of the sine wave m(t) f, is the sampling rate Ju #5 the frequency of the sine wave. For k = 4 mV and k = 60 mV, does the slope overload occurs ? If so, in which case? Given, m{t) = 0.1 sin (2x 1034)aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.Digital Communications 2-13 Delta Modulation Putting for A,, from equation 2.3.6, x a 23.7) © BART? “ This is an expression for signal power in delta modulation. Gi) To obtain noise power a6 We know that the maximum quantization error in delta modulation is equal to step size 8. Let the quantization error be uniformly distributed over an interval [-6,5} This is shown in Fig. 2.3.4 From this figure the PDF of quantization error can be expressed as, Fig. 2.3.4 Uniform distribution of quantization error 0 for <8 fe) = x for ~88 The noise power is given as, V2, Noise power = —moise Here V2,,,, is the mean square value of noise voltage. Since noise is defined by random variable '' and PDF /, (e), its mean square value is given as, mean square value = E[e?] mean square value is given as, Ee?] = fers (eydeaa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.Digital Communications 2-17 Modulation * * B~ 200«103 °° (i) To obtain step size From equation 2.3.4 we have, 8 Am OapT, Under this condition slope overload will not occur. From above equation step size will be, & 2 2nfpTAy Putting values in above equation, 1 —*__x 200x103 2 0.157 V . 8 > 2nx10,000x 0.5 Thus the step size greater than 157 mV will prevent the slope overload. {ii) To obtain signal to noise ratio Signal to noise ratio of delta modulation system is given by equation 2.3.12 as, Ss 3 N ~ sewp2r3 This is post filtered signal to noise ratio. In this example value of 'W’ is not given. Hence we will calculate signal to noise ratio from equation 2.3.7 and equation 2.3.10 as, 2 = Be hate oR 3s =— 2. 8x? fAT? Zin Putting values in above equation. s 3 sn? x (10,000) N 1 (200x103)? = 152 =118 dB.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.24 Delta Modulation Digital Communications — 1 }o21,_if?]! X? = f PA (adx= fx?-Far= 5] -1 at 3 x2 se Normalized signal power = = =X? with R = 1 =i a9 Ww Hence signal to noise ratio becomes, 1 S$ _ Signalpower_ 33 155 N ~ Noise power 215x102 or (3) = 10log 9 155 = 21.9 dB NJ ap Theory Questions 1. Explain delta modulation in detail suitable diagram. Explain ADM and compare its performance with DM. 2. What is slope overload distortion and granular noise in delta modulation and how it is removed in ADM ? Unsolved Example 1. What is the maximum power that may be transmitted without slope overload distortion ? [Ans. e 1 _bnfaTy 2.4 Comparison of Digital Pulse Modulation Methods Table 2.4.1 shows the comparison of PCM, Differential PCM, Delta Modulation and Adaptive Delta Modulation. The comparison is done on the basis of various parameters like transmission bandwidth, quantization error, number of transmitter bits per sample etc.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.Digital Communications 3-3 Digital Modulation Techniques 1, Message source : It emits the symbol at the rate of T seconds. 2. Encoder : It is signal transmission encoder. It produces the vector s, made up of 'N’ real elements. The vector s; is unique for each set of 'M' symbols. 3. Modulator : It constructs the modulated carrier signal s,(t) of duration 'T seconds for every symbol m,, The signal si(t) is energy signal. 4, Channel : The modulated signal s(t) is transmitted over the communication chanrel. * The channel is assumed to be linear and of enough bandwidth to accommodate the signal s,(t). * The channel noise is white Gaussian of zero mean and psd of 2. 5. Detector : It demodulates the received signal and obtains an estimate of the signal vector. 6. Decoder : The decoder obtains the estimate of symbol back from the signal vector. Here note that the detector and decoder combinely perform the reception of the transmitted signal. The effect of channel noise is minimized and correct estimate of symbol 1it is obtained. 3.2 Binary Phase Shift Keying (BPSK) 3.2.4 Principle of BPSK © In binary phase shift keying (BPSK), binary symbol ‘1’ and ‘0' modulate the phase of the carrier. Let the carrier be, s() = Acos(2nfyt) «= G21) ‘A’ represents peak value of sinusoida! carrier. In the standard 10 load register, the power dissipated will be, = 1 = P= 5A A = \2P ++ (3.2.2) * When the symbol is changed, then the phase of the carrier is changed by 180 degrees (n radians). i © Consider for example, Symbol '1' = 5; () = V2P cos(2n fy!) was (3.2.3)aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.Digital Communications 3-7 Digital Modulation Techniques =b) E [1 +cos 2(2m fy #+6)] =» (82.8) 6) Bit synchronizer and integrator : The above signal is then applied to the bit synchronizer and integrator. The integrator integrates the signal over one bit period. The bit synchronizer takes care of starting and ending times of a bit. * At the end of bit duration T,, the bit synchronizer closes switch 52 temperorily. This connects the output of an integrator to the decision device. It is equivalent to sampling the output of integrator. * The synchronizer then opens switch $, and switch 51 is closed temperorily. This resets the integrator voltage to zero. The integrator then integrates next bit. * Let us assume that one bit period ‘7,’ contains integral number of cycles of the carrier. That is the phase change occurs in the carrier only at zero crossing. This is shown in Fig. 3.2.1 (0). Thus BPSK waveform has full cycles of sinusoidal carrier. To show that output of integrator depends upon transmitted bit © In the K*# bit interval we can write output signal as, Sg (ET,) = very fe t [14 cos 2(2m fy t+8)] dt (e-1) Tp from equation 3.2.8 The above equation gives the output of an interval for k!! bit. Therefore integration is performed from (k -1)T,, to kT,. Here T,, is the one bit period. * We can write the above —_ as, kT Sy (KT,) = very B| ; ldt+ [cos 2(2mfyt +0) dt (1) Ty (DT kT Here {cos 2(2nfo t+ 0) dt=0, because average value of sinusoidal waveform is (k-1) Ty zero if integration is performed over full cycles. Therefore we can write above equation as, 89 (kT,) = vary fe r ldt (DTaa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.| Communications 3-11 Digital Modulation Techniques Interchannel Interference and ISI : Let's assume that BPSK signals are multiplexed with the help of different carrier frequencies for different baseband signals. Then at any frequency, the spectral components due to all the multiplexed channels will be present. This is because S(f) as well as Sppsx (f) of every channel extends over all the frequency range. * Therefore a BPSK receiver tuned to a particular carrier frequency will also receive frequency components due to other channels. This will make interference with the required channel signals and error probability will increase. This result is called Interchannel Interference. * To avoid interchannel interference, the BPSK signal is passed through a filter.This filter attenuates the side lobes and passes only main lobe. Since side lobes are attenuated to high level, the interference is reduced. Because of this filtering the phase distortion takes place in the bipolar NRZ signal, ie. b(). Therefore the individual bits (symbols) mix with adjacent bits (symbols) in the same channel. This effect is called intersymbol interference or ISI. «The effect of ISI can be reduced to some extent by using equalizers at the receiver. Those equalizers have the reverse effect to that filter's adverse effects, Normally equalizers are also filter structures. 3.2.5 Geometrical Representation of BPSK Signals We know that BPSK signal carries the information about two symbols, Those are symbol ‘I’ and symbol ‘0’. We can represent BPSK signal geometrically to show those two symbols. (i) From equation 3.2.6 we know that BPSK signal is given as, s() = b(t)-V2P cos (2n fy t) wee (3.2.15) (i) Let's rearrange the above equation as, s() = b@JPT, - Fe conf!) w= (3.2.16) ae cos (2n fo t) represents an orthonormal carrier signal. Equation b 3.2.14 also gives equation for carrier. It is slightly different than 9, (f) defined here. Then we can write equation 3.2.16 as, si) = b(OYPT, 1.0 ww» 3.2.17)aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.1 Communications 3-15 Digital Modulation Techniques intevatNo.of 7 | 2 | 3 eoare.O jf ane Po. oye ype wT) Fig, 3.3.2 DPSK waveforms From the waveforms of Fig. 3.32 it is clear that b(t-T,,) is the delayed version of b( by one bit period T,. The exclusive OR operation is satisfied in any interval ie. in any interval b (8) is given as, bw) = d@@b¢-T,) » 83.1) While drawing the waveforms the value of b(!-T;) is not known initially in interval no. 1. Therefore it is assumed to be zero and then waveforms are drawn. Important conclusions from the waveforms 1. Output sequence b(t) changes level at the beginning of each interval in which d(t)=1 and it does not changes level when d(t) = 0. Observe that d (3) =1, hence level of b (3) is changed at the beginning of interval 3. Similarly in intervals 10, 11, 12 and 13 d ()=1. Hence b (!) is changed at the starting of these intervals. In interval 8 and 9 d(f)=0. Hence b(t) is not changed in these intervals. 2. When d()=0, bQ)=b(t-Tp) and When d ()=1, b=b0-T,) 3. In interval no. 1. we has assumed b(!-T;,)=0 and we obtained the waveform as shown in Fig. 3.3.2. If we assume b(t-T,)=1 in interval no. 1, then the waveform of b(t) will be inverted. But still b(t) changes the level at the beginning each interval in which d(t) = 1. 4. The sequence b(!) modulates sinusoidal carrier.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.Digital Communications 3-19 Digital Modulation Techniques and PTy, then d(f +PTp, then d(t)= If 55 (kT) = { 3.3.2 Bandwidth of DPSK Signal We know that one previous bit is used to decide the phase shift of next bit Change in b(t) occurs only if input bit is at level ‘1’. No change occurs if input bit is at level ‘0’. Since one previous bit is always used to define the phase shift in next bit, the symbol can be said to have two bits. Therefore one symbol duration (T) is equivalent Ito two bits duration (27,). ie. Symbol duration T = 21, G31) Bandwidth is given as, 2 BW = = -~ i 7 or BW = f, .- (3.3.12) Thus the minimum bandwidth in DPSK is equal to f, ; ie. maximum baseband signal frequency. 3.3.3 Advantages and Disadvantages of DPSK DPSK has some advantages over BPSK, but at the same time it has some drawbacks. Advantages : 1) DPSK does not need carrier at its receiver. Hence the complicated circuitry for generation of local carrier is avoided. 2) The bandwidth requirement of DPSK is reduced compared to that of BPSK. Disadvantages : 1) The probability of error or bit error rate of DPSK is higher than that of BPSK. 2) Since DPSK uses two successive bits for its reception, error in the first bit creates error in the second bit. Hence error propagation in DPSK is more. Whereas in PSK single bit can go in error since detection of ~sch bit is independent.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.Digital Communications 3-23 Digital Modulation Techniques Step 2 : Demultiplexing into odd and even numbered sequences The demultiplexer divides b(t) into two separate bit streams of the odd numbered and even numbered bits. b, (f) represents even numbered sequence and b, (f) represents odd numbered sequence. The symbol duration of both of these odd and even numbered sequences is 2T,. Thus every symbol contains two bits. Fig. 3.4.2 (b) and () shows the waveforms of b, (f)andb, (t). Observe that the first even bit occurs after the first odd bit. Therefore even numbered bit sequence b, (f) starts with the delay of one bit period due to first odd bit. Thus first symbol of b, (t) is delayed by one bit period 'T,,' with respect to first symbol of b, (f). This delay of T), is called offset. Hence the name offset QPSK is given. This shows that the change in levels of b, (f) andb, (#) cannot occur at the same time because of offset or staggering. Step 3 : Modulation of quadrature carriers The bit stream b,(t) modulates carrier /P, cos (2x fy !) and 5, (!) modulates P sin(2n fy). These modulators are balanced modulator. The two carriers 2, cos (2m fy Nand JP, sin (2m fy f) are shown in Fig. 3.4.2 (d) and (¢). These carriers are also called quadrature carriers. The two modulated signals are, 5, (f) = b, (JP; sin (2m fy) -- G41) and Sy () = by OP, cos (2m fy t) ww. (3.4.2) Thus s, ()ands, (t) are basically BPSK signals and they are similar to equation 3.23 and equation 3.25. The only difference is that T=27, here. The value of b, (®andb, (f) will be +1V or ~1V. Fig. 3.4.2 (f) and (g) shows the waveforms of s, ands, (0). Step 4: Addition of modulated carriers The adder of Fig. 3.4.1 adds these two signals b, (t) and b, (0). The output of the adder is OQPSK signal and it is given as, si) = s, +s, 0 = by (W) VP; cos (2m fo t) +b, (t) YP, sin (2m fot) w= (3.4.3) Step 5 : QPSK signal and phase shift Fig. 342 (h) shows the QPSK signal represented by above equation. In QPSK signal of Fig. 3.4.2 (h), if there is any phase change, it occurs at minimum duration of T). This is because the two signals s, (t) ands, (t) have an offset of 'T,’. Because of this offset, the phase shift in QPSK signal is 3 It is clear from the waveforms of Fig. 3.4.2 that b, (!) andh, (f) cannot change at the same time because of offset between them. Fig. 3.4.3 shows the phasor diagram of QPSK signal of equation 3.4.2.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.Digital Communications 3-27 jital Modulation Techniques To show that output of integrator depends upon respective bit sequence. + * Let's consider the product signal at the output of upper multiplier. s(t) sin (2m fy t) = bg (8) JP, cos(2n fo t) sin (2n fy t)+b, (JP, sin? (2nfy) —... G44) * This signal is integrated by the upper integrator in Fig. 3.4.4. (2k+1) Th (2k+1) Th il s(t) sin (2m fo t) dt = b, (t) JP, f cos (2m fy t) sin (2m fa t) dt (2k-1) Ty (2k-1) Tp, (241) Th +b, (0 YP, J sin? (2m fy t) dt QED), Since 5 in (22) ie ginxvoune mae 2 5 (2x) and sin? (x) = 5 [1 ~cos (2a) + Using the above two trigonometric identities in the above equation, (2k+1) Ty bf 2D Te bo (2k+1) Tp J s (#) sin (2n fy 8) dt abo VE sinAnfy tdt+ e 1-dt Ok-D Ty Qk-1) Th, Qk-1) Ty 6 ) fe oD ORL J cosdnfotdt (24-1) T% * In the above equation, the first and third integration terms involves integration of sinusoidal carriers over two bit period. They have full (integral number of) cycles over two bit period and hence integration will be zero. (2k+1) Th b. OJP J s@sin(anfohat = -£ i DR wee} 2 (2-1) Th bP, OE y 2, = bP, T, ws (3.4.5) * Thus the upper integrator responds to even sequence only. Similarly we can obtain the output of lower integrator as b, (t)aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.Digital Communications 3-31 Digital Modulation Techniques Thus the length of each signal point from origin is JE, . * We know that b, () and, (#) represent two successive bits. There is an offset of 'T,’ between b, ()andb, (f). Therefore b, (f)andb, (t) both cannot change their levels simultaneously. Therefore either b, (t) orb, (8) can change at a time. * Let's say that b, (t) =b, (t)=1 representing signal point ‘A’ in Fig. 3.46. In the next bit interval if b, (!)=~1, then signal point will be 'D'. Otherwise if b, (1) changes its level (ie. b, (t)=-1), then next signal point will be 'B'. Thus from signal point 'A', then next signal points will be either ‘D' or 'B’. Distance between signal points : Normally the ability to determine a bit without error is measured by the distance between two nearest possible signal points in the signal space. Such points differed in a single bit. For example signal points 'A’ and 'B' are two nearest points since they differ by a single bit b, (1). As ‘A’ and ‘B' becomes closer to each other, the possibility of error increases. Hence this distance should be as large as possible. This distance is denoted by ‘d’, In Fig. 3.4.6, the distance between signal points ‘A’ and ‘B' is given as, a = (JE)? +(JE)? By d= f2 wns (3.4.18) or d = 2JP,T, =2JE, a. (3.4.19) Compare this distance with the distance of BPSK signals given by equation 3.2.20. This shows that the distance for QPSK is the same as that for BPSK. Since this distance represents noise immunity of the system, it shows that noise immunities of BPSK and QPSK are same. 3.4.3 Spectrum of QPSK Signai Step 1 : PSD of NRZ waveform The input sequence b(}) is of bit duration T,,. It is NRZ bipolar waveform. In section 3.2.4 we have obtained the power spectral density of such waveform as, . 2 si) = v2T, ner from equation 3.2.12 and V, =./P,, then above equation becomes, so ~ nn (sgt The above equation gives power spectral density of signal b(!).aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.Digital Communications 3-35 Digital Modulation Techniques te = — ty PeltNB, sin(2nfot) Ps sib than ~~ @Psk signet f TY EV L(g) sit)=s,(0 + ot (@) st0=540*540) ee NT 1 ' T sdb Sted ad PPh Phase shifts of 5 pf" Fig. 3.4.8 QPSK waveformaa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.Digital Communications 3-39 jital Modulation Techniques The above two equations are orthonormal waveforms. Fig. 3.5.1 shows the si space diagram based on equation 3.5.6. The orthonormal carriers >, (!) andé, (f) form two axes. The signal points So,5;,5q....Sq,1 ate placed on the circumference of the circle, The signal points are equispaced with the phase shift of 2%. The distance of each signal point from the origin is /P, T,- 8} ont. FE .c05(2afgt) Fig. 3.5.1 Signal space diagram or geometrical representation of M-ary PSK signals Here PT, = E, (Symbol energy) = 85.9) Thus we can say that QPSK is the special case of M-ary PSK with M=4. Then the signal space diagram of QPSK and 4-ary PSK will be similar 3.5.2 Power Spectral Density of M-ary PSK PSK and QPSK are the special cases of M-ary PSK. The symbol duration for M-ary PSK is given by equation 35.2 as, T, = NT, w+ (3.5.10) Here N is the number of input successive bits combined. The baseband power spectral density of QPSK is given as, s 2 Spcopsx) () = 2P,T; ao] from equation 3.4.23 If we put T, =NTj, in above equation we will get power spectral density of M-ary PSK i.e., _ ‘sin(nf NT,) Sp(f) = 2P.NTy ae s» (35.11)aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.Digital Communications 3-43 Digital Modulation Techniques Meaty PSK signal s(t) Raise input to Bit i sequence M" power Parallel Ye serial s(t) sin(2ntgt) cos(2nfgt) ‘sin(2rfot) Fig. 3.5.5 M-ary PSK receiver A/D converter, which reconstructs 'N' bit symbol. This 'N’ bit symbol is given to the parallel to serial converter. It then generates the bit sequence b(t). ma Example 3.5.1: A ary PSK has the transmitted waveforms, in si) = tea(2mserF) «=» G5.15) i = 0,1,2,3and0 Example 3.5.5: Derive an expression for the spectral spread of 16-ary PSK system. Solution : Power spectral density of M-ary PSK is given by equation 3.5.11 as,aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.Digital Communications 3-51 Digital Modulation Techniques VP ,cost2ntat) VP, sin(2r)) Fig. 3.6.2 Generation of QASK signal digital to analog converter and b,, andb,, 3 are applied to lower digital to analog converter. Depending upon two input bits, the output of digital to analog converter takes four output levels. Thus A, (t)andA,() takes 4 levels depending upon combination of two inputs bits. A, (f) modulates the carrier .[P, cos(2n fy #) and A,() modulates ,/P, sin(2r fy). The adder combines two signals to give QASK signal. It is given as, s(t) = A,() JP, cos(2nfy t) + Ag) JP, sin(2n fy t) ve. (3.6.11 (a)) If we compare the above equation with equation 3.6.11 We can write A,@)andA,() = +02 or +3V02 we (3.6.12) (depending upon input to D/A converter) 3.6.2.2 Receiver of QASK Signal Fig. 3.63 shows the receiver of 16-QASK (4bits QASK) system. The input signal s() is raised to 4" power. It then passed through a bandpass filter centered around the frequency 4fp the signal is then divided in frequency by four. It gives a coherent carrier cos (27 fy t). Quadrature carrier sirt 2x fy t) is produced by phase shifting of 90° The inphase and quadrature coherent carriers are multiplied with QASK signal s (8). Since the amplitudes of A,()) and A, (f) are bit constant and equal, let us check whether we can really recover the carrier correctly. The 4” power QASK signal is, st(t) = P2[A,(@cos(2nfy t) +A (0) sin (2n fo 1" «» 3.6.13)aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.Digital Communications 3-55 Digital Modulation Techniques 3.7.1 BFSK Transmitter From the table 3.7.1, we know that P), () is same as b(t). And P, (f) is inverted version of b(t). The block diagram of BFSK transmitter is shown in Fig. 3.7.1. We know that input sequence b(t) is same as P;, (1). An inverter is added after b(f) to get P, (#. Py (and P, (t) are unipolar signals. The level shifter converts the ‘+1’ level to JP, T,. Zero level is unaffected. Thus the output of the level shifters will be either JP. T, (if ‘+1') or zero (if input is zero). Further there are product modulators after level shifter. The two carrier signals >, (f) and$, (f) are used. 6, (t) and, (f) are orthogonal to each other. In one bit period of input signal (i.e. T,), 9; (t) or 92 (f) have integral number of cycles. = Pecos bit) BFSK Input —>| signal sequence s(t) VPSTEPLO Fig. 3.7.1 Block diagram of BFSK transmitter Therefore the modulated signal has continuous phase. Such BFSK signal is shown in Fig. 37.2. The adder then adds the two signals. sft) st) s(t) soft) si(t) s(t) Fig. 3.7.2 BFSK signalaa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.Digital Communications 3-59 wee [Comparaigg>>——e b(t) signal Fig. 3.7.5 Biock diagram of BFSK receiver The outputs of filters are applied to envelop detectors. The outputs of detectors are compared by the comparator. If unipolar comparator is used, then the output of comparator is the bit sequence b (\). 3.7.5 Geometrical Representation of Orthogonal BFSK or Signal Space Representation of Orthogonal BFSK Orthogonal carriers are used for M-ary PSK and QASK. The different signal points are represented geometrically in >, 65 plane. For geometrical representation of BFSK signals such orthogonal carriers are required. From Fig, 3.7.1, we know that, two carriers $, (t) and, (f) of two different frequencies f,, andf, are used for modulation. To make 6 (f) and, (t) orthogonal, the frequencies f;,; and f, should be some integer multiple of base band frequency 'f,. ie fa = fy vee (37.14) and fl = fy wee (3.7.15) Here f, ==, then the carriers will be o) = tr cos (2n m fy #) -. @.7.16) and 2) = EE wenm 5 vs G.7.17)aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.Digital Communications 3-63 Digital Modulation Techniques 3.8.1.2 Receiver Fig. 3.8.2 shows block diagram of M-ary FSK receiver. It is the extension of BFSK receiver of Fig. 3.8.1. The M-ary FSM signal is given to the set of ‘M’ bandpass filters. The center frequencies of those filters are fy, fi, fo/.--fy-1- These filters pass their particular frequency and alternate others. The envelope detectors outputs are applied to a decision device. The decision device produces its output depending upon the highest input. Depending upon the particular symbol, only one envelope detector will have higher output. The outputs of other detectors will be very low. The output of the decision device is given to ‘N’ bit analog to digital converter. The analog to digital converter output is the ‘N’ bit symbol in parallel. These bits are then converted to serial bit stream by parallel to serial converter. In some cases the bits appear in parallel. Then there is no need to use serial to parallel and parallel to serial converters. Penchass Envelope detector = Envelope Mary detector FSK signal | Fig. 3.8.2 Block diagram of M-ary FSK system 3.8.2 Power Spectral Density and Bandwidth of M-ary FSK We know that for M symbol fo, fz ---fm-1 frequencies are used for transmission. The probability of error is minimized by selecting those frequencies such that transmitted signals are mutually orthogonal. If those frequencies are selected as successive even harmonics of symbol frequency f,, then transmitted signals will be orthogonal. Let’s say that the lowest carrier frequency fo is the k” harmonic of symbol frequency ie., fo = - B81) Then the other frequencies will be, fr = K+Df fy =k44) f «.. ete. vs G82)aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.Digital Communications Bt sequence wt) Bit sequence bolt) Bit sequence b(t) oyoon 3 nueosdit +9) fe » (1.25+0.28}f, = fy or) 2 os =(1.25+0.25)f, =1 Of b, Yrasx(t) Fig. 3.9.1 MSK waveforms (a) Bipolar NRZ waveform representing bit sequence (b) Odd bit sequence waveforms b, (t) (c) Even bit sequence waveform b, (t) f, (4) Wavoforms of frequency “2 used for smoothing of by (t)and by (t) (e) Modulating waveform of even sequence (f) Modulating waveform of odd sequence (g) Waveform of frequency fy, {h) Waveform of frequency f, (i) MSK waveformaa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.Digital Communications 3-71 Digital Modulation Techniques MSK is called “Shaped QPSK” 7 In QPSK, b(t) and b,() directly multiply the carrier. Hence there are abrupt changes in phase (and hence amplitude) in the QPSK waveform. In MSK, two waveforms b,(t) sin (2nt / 4T),) and b,(t) cos(2xt / 4T,) are first generated as shown in Fig. 3.9.1 (¢) and (f). These waveforms multiply the carriers. Thus b,(t) sin (2nt / 4T)) and b(t) cos (2xt / 4T,) does not have abrupt changes in their amplitudes. Hence the multiplied carriers have no abrupt changes and they have continuous phase. The MSK waveform of Fig. 3.9.1 is drawn for m=5. From equation 3.9.14, we can obtain the carrier frequency as, fo = Zhe =3h, =13 f, b,(Osin2nit / 4T,)) f= 1.25f, YP; [bg(t)sin2n(t /47,)}oos(2xt) belthoos2n(t / 47,)| fo = 1.25, VAP; [dplt}cos2n(t/ 4Tp)Isin(Zatot)) Fig. 3.9.2 MSK waveforms showing ‘smoothing’ effect on modulated carriers “shaped QPSK". {a) & (d) smoothed modulating waveforms (Odd & Even sequences) (b) & (0) Carrier fy =1.25f,, (m=5) (c) & (f) Modulated carriersaa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.Digital Communications 3-79 Digital Modulation Techniques 3.9.5 Advantages and Disadvantages of MSK as Compared to QPSK From the discussion of MSK we can now compare the advantages of MSK over QPSK. Advantages : 1. The MSK baseband waveforms are smoother compared to QPSK. 2. MSK signal have continuous phase in all the cases, whereas QPSK has abrupt phase shift of Form. 3. MSK waveform does not have amplitude variations, whereas QPSK signal have abrupt amplitude variations. 4. The main lobe of MSK is wider than that of QPSK. Main lobe of MSK contains around 99% of signal energy whereas QPSK main lobe contains around 90% signal energy. 5. Side lobes of MSK are smaller compared to that of QPSK. Hence interchannel interference is significantly large in QPSK. 6. To avoid interchannel interference due to sidelobes, QPSK needs bandpass filtering, where as it is not required in MSK. 7. Bandpass filtering changes the amplitude waveform of QPSK because of abrupt changes in phase. This problem doesnot exist in MSK. The distance between signal points is same in QPSK as well as MSK. Hence the probability of error is also same. However there are few drawbacks of MSK, Disadvantages : 1. The bandwidth requirement of MSK is 1.5 f,, whereas it is f, in QPSK. Actually this cannot be said serious drawback of MSK. Because power to bandwidth ratio of MSK is more. 99% of signal power can be transmitted within the bandwidth of 1.2 f, in MSK. While QPSK needs around 8 f, to transmit the same power. 2. The generation and detection of MSK is slightly complex. Because of incorrect synchronization, phase jitter can be present in MSK. This degrades the performance of MSK. 3.9.6 Gaussian MSK Power spectra of MSK is given in Fig. 3.9.8, Observe that the the main lobe is wide. This makes MSK unsuitable for the applications where extremely narrow bandwidths and sharp cut-offs are required. Slow decay of MSK psd curve creates adjacent channel interference. Hence MSK cannot be used for multiuser communications. This problem can be overcome with Gaussian MSK. Fig. 3.9.8, shows the little modification.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.Digital Communications 3-83 Digital Modulation Techniques ‘Symbol ‘0’ ‘Symbol ‘1" + 2 ont } PE (0) Fig. 3.10.2 Signal spaco diagram of ASK Therefore the distance between the two signal points will be, d= JPT,=JE, «+ 3.10.3) 3.10.2 Generator and Detector of ASK 3.10.21 ASK Generator Fig. 3.10.3 shows the ASK generator. The input binary sequence is applied to the product modulator. The product modulator amplitude modulates the sinusoidal carrier. It passes the carrier when input bit is ‘1’. It blocks the carrier (i.e. zero output) when intput bit is ‘0’. The wavefrom of ASK is as shown in Fig. 3.10.1. Bi Binary ASK signal Product . signal tt) Modulator xt) Cartier BP, cos(2nfg!) Fig. 3.10.3 Block diagram of ASK generator 3.10.22 ASK Detector Fig. 3.10.4 shows the block diagram of coherent ASK detector. The ASK signal is applied to the correlator consisting of multiplier and integrator. The locally generated coherent carrier is applied to the multiplier. The output of multiplier is integrated over one bit period. The decision device takes the decision at the end of every bit period. It compares the output of integrator with the threshold. Decision is taken in favour of ‘I’ when threshold is exceeded. Decision is taken as ‘0’ if threshold is not exceeded.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.Data Transmission The analog signal is converted to digital or binary waveform by means of waveform coding techniques. In first and second chapter we have seen such waveform coding techniques. They are PCM, DM, ADM, DPCM etc. This digital data is then converted to RZ, NRZ, AMI etc. type of signal waveforms. The digital (binary) signal then can be transmitted either using baseband transmission or using bandpass transmission. In bandpass transmission, the digital signal modulates high frequency sinusoida’ carrier. The analysis of such techniques we have seen in previous chapter. They are called digital modulation techniques. With the help of such techniques, it is possible to transmit data over long distances. In baseband transmission, the data is transmitted without modulation. During the transmission of data over the channel, it is corrupted by noise. Hence at the receiver, the noisy signal is received. Therefore correct detection of the transmitted signal is difficult. For example consider the transmitted signal and received noisy signal as shown in Fig. 4.1 (a) and (b). The received signal %(') is a noisy signal at the receiver. Let us consider that, the detector checks i(#) at ‘I’ during every bit interval. In above figure observe that the decision in first interval will be correct ie. symbol 'l'. But in second interval, the decision will be 'l' but it is wrong. At the time when detector checks %() [ie. at !=T], noise pulse is detected and decision is taken in favour of ‘1’. But actually symbol '0' is transmitted in second interval as shown in Fig. 4.1(a). Thus errors are introduced because of noise. The detecting method of the baseband signal perform following jobs: (i) The detection method should attenuate noise and amplify signal, i.e. it should improve signa! to noise ratio of the received signal. (ii) The detection method should check the received signal at the time instant in the bit interval when signal to noise ratio is maximum. (iii) The detection should be performed with minimum error probability. In this chapter we will study some methods for detection of digital signals. We will also compare these methods on the basis of their performances. (4-1)aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.Digital Communications 4-7 Data Transmission doe EC Let us rearrange the above equation as, — yo 2) 2 no = No aa (2) tix 4 xT t dx= : du and limits will be unchanged, therefore above equation becomes, Xo ae a du . Not F sinuy” ay ~ ee i Since the function £"™ ig squared, we can waite above equation as, a _ NoT yf (sinu)* ny = raf (54) du oO Ont’ u NoT ne? Not 2x? 5 By equation C-46 in appendix 'C’ # ne) = + (41.9) The above relation gives noise power at the output. We obtain the signal to noise power ratio at output of integrator as, Signal power = Noise power Putting the values of signal power from equation 4.1.3 and noise power from equation 4.19 we get, Signal to noise ratio, Arr? Zz = ° = Not 2x?aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.Digital Communications 4-11 Data Transmission Thus as shown in Table 4.1.1, the error is introduced depending upon probability that 1g (0) takes a particular value. These probabilities can be obtained from PDF of 1g (). We know that the probability density function (PDF) of the gaussian distributed function is given by standard relation as, 1 2) 202 eee (4m */ 20! we (4.1.15) fx @ Par ¢ ) Here fy (x) is the PDF of random function x. mis the mean value and G is the standard deviation. Here since we want to evaluate PDF for white gaussian noise we have, x= no Since this noise has zero mean value, m=0 equation 4.1.15 can be written as, fic (mg 0) = = ev Pry ti2/202 wn (4.16) oV2n The standard deviation o is given as, 1 © = [mean square value ~ square of mean value]2 — 1 ie. o, = [X?-m2]2 x Here mean square value X? .. from equation 4.1.9 And of mean value m, =0 for this noise. ——1 NOT (n§ OP = yoo 6 Hence equation 4.1.16 can be written as, 1b wr?/: (ar) ——— 2 Not jo an 22 fx (9 0) Here note that mf) is the function like ‘y. It is random variable and we are evaluating its PDF. On simplifying above equation we get,aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.aa You have either reached a page that is unavailable for viewing or reached your viewing limit for this book.Digital Communications 4-15 Data Transmission Solution : Given data: psd of white noise, N0=10°° w/Hz amplitude, A = 10x10 v data rate = 10103, Hence Tp =——+—. 10x10 Probability of occurrence of both the symbols is equal, i.e. 05. @ To obtain probability of error P, Equation 4.1.22 give probability of error of integrate and dump receiver as, 1 |A2T Pe = 5 ame a Putting values in above equation, 2 3 1 (10x10) 1 = erfe , Here T=-———— ze 2x10 x10x10> "10x10 h erfe v5 Pe W Gi) To obtain 'A' for bit rate of 10 Mbits/sec The probability of error is to be maintained same. i.e., 1 1 Az 1 sefevS = safc |——_____._, Peo 2 2 2x10 x10x10° 10x10° 2 5= wet __ 2x10 x 10x 10° A = V0 = 03162 volts Thus the amplitude must be increased to 0.3162 volts to maintain same probability of error. Theory Questions 1. Explain how integrator is used to detect baseband digital signals. Derive the expression for signal to noise ratio of integrate and dump receiver. 2. Derive an expression for error probability of an integrator.