kL Pte MOC taceePulse Digital Modulation Introduction ‘+ There are three types of modulation () Amplitude modulation (ii) Angle modulation (iii) Pulse modulation «Pulse modulation can be further classified as, (Pulse analog modulation (ii) Pulse digital modulation + The above two techniques can be further classified as, (i) Pulse ampftude modulation () Pulse code modulation (i) Pulse position meduiation (@) Dota modulation (ii) Pulse duraton moduiation (@) Adoptive deta modulation ((¥) Differential pulse code modulation ‘+ In the above techniques following points are studied : (Principle of operation (i) Transmitter and receiver block diagram (ii) Error analysis (iv) Signal to quantization noise ratio. (1-4)Digital Communications: 41-2 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 circuils 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. Internet 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. 1.21 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 aud transmission rate are normally used as performance measures of the digital communication system, Fig. 1.21 Basic dialtal communication system 1.2.1 Information Source The information source generates the message signal to be transmitted. In case of analog communication, the information sovrce is analog. In case of digital communication, the information source produices 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 14 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 diserete 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. 4) Probabilistic dependence of symbols in a sequence : The information carrying capacity of cach 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 > Source entropy Gits/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 (ie. 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 28 = 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 Communications 155 Pulse Digital 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 = B0bits/sec 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. 4) 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 (i.e. 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 b9 by be _b % 110 ° i400 o10 1 oto 4 ooo o oooo 1 1444 tad ‘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 recurrently after the block of few symbols or added in every symbol as shown in Table 1.21. 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. ©) Coding efficiency, which is the ratio of data rate at the input to the data rate at the output of encoder. @) Error control capabilities, ie. detecting and correcting errors ) 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 sz(t) to transmit binary ‘1’. For example consider the output of digital modulator shown in Fig. 1.22. femfear—f os fs foo fs fo fo | a oat Fig. 1.2.2 Frequency modulated output of a digital modulator The signal s,() has low frequency compared to signal s2(0). 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 5;(t) and 52(f) 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. ©) Synchronous or asynchronous method of detection and 4) Complexity of implementation. 1.2.5 Communication Channel As we have seen in the preceding sections, the connection between transmitter and receiver is established tvough 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 camry 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. b) Signal attenuation : Ik occurs due to intemal resistance of the channel and fading of the signal. © 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 - @) 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 :Communication Pulse Digital Modulation a) Very large bandwidths are possible. b) Transmission losses are very small. ©) Electromagnetic interference is absent. ) They have small size and weight. €) 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 7 + ACT signal cannot be processed in the digital processor or computer. + To enable digital transmission of CT signals. Fig. 13.1 shows the CT signal and its sampled DT signal. In this figure observe that the CT signal is sampled at t = 0, T,, 2T,, 3T,, ... and so on. Eg Fig, 1.3.1 CT and its DT signalDigitat communications 1-10 Pulse Digital Modulation ‘* Here sampling theorem gives the criteria for spacing ‘T,’ between two successive samples. + The samples x5(!) must represent alll the information contained in x(t). ‘The sampled signal x5() is called discrete time (DT) signal. It is analyzed with the help of DTFT and z-transform. 1.3.2 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 bigher than W Hert, ts completely described by specifying the values of the signal at instants of time separated by ay uy 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. ie, f= Ww Here f, is the sampling frequency and W is the higher frequency content Proof of sampling theorem There are two parts : (I) Representation of »() in terms of its samples (0) Reconstruction of x(t) from its samples. Part I: Representation of x) in its samples 37,) Step.1: Define x() Step 2: Fourier transform oF xsl i ie x30 ‘Step 3: Relation between X(/) and Xs) Step 4: Relation between x(#) and (#7)Digital Communications 111 Pulse Digital Modulation Step 1: Define x9(t) Refer Fig. 1.3.1. The sampled signal x9(f) is given as, xs) = Yx08¢-n7,) +: 03.1) Here observe that xs(t) is the product of xs and impulse train 6(f) as shown in Fig. 1.3.1. In the above equation 8(t~nT;) indicates the samples placed at'+T;, +27, £37; ... and so on, Step 2: FT of xs(t) i.e. X5(0 ‘Taking FT of equation (1.3.1). X(N = =| Sem = FF (Product of a() and impulse train) We know that FT of product in time domain becomes convolution in frequency domain. ie. Xa) = FT ())*FT(8(t-n7,)) (1.32) By definitions, x(f) 2+ X(f) and Bt-nt,) 22 f, FE-nh) Hence equation (13.2) becomes, Xa = XO D8F-nh) tice Since convolution is linear, Xe = & ExQesg-mh) =f Exg-m By shifting property of impulse furiction Se Xf ~2fe) + fe XU —fe)+ Se XU) + fe XU fe) the XPD fe) teDigital Communications 1-12 Pulse Digital Modulation Comments @) The RHS of above equation shows that X(f) is placed at $f, 42f,,43f,,- (i) This means X(f) is periodic in fy (ii) If sampling frequency is f, = 2W, then the spectrums X(j) just touch each other. “OZ ZANE ZINUZT I me ew | |W | wT & Tew] 2 | Clee SEPP er oP —f=nt,-| Fig, 1.3.2 Spectrum of original signal and sampled signal Step 3 : Relation between X(/) and Xg(/) Important assumption : Let us assume that f, = 2W, then as per above diagram. Xa = EX for-W Sf 2W ‘When the sampling rate is made higher than 2W, then the spectrums will not overlap and there will be sufficient gap between the individual spectrums. This is shown in Fig. 1.35.Digital Communications 1-17 Pulse Digital Modulation ‘The sampling rate is, f, = 2W. Ideally speaking there should be no aliasing. But there can be few components higher than 2W. These components create aliasing. Hence a lowpass filter is used before sampling the signals as shown in Fig. 1.3.6. Thus the output of lowpass filter is strictly bandlimited and there are no frequency components higher than ‘W’. Then there will be no aliasing, XQ Fig. 1.3.6 Bandlimiting the signal. The bandlimiting LPF Is called prealins filter 1.3.4 Nyquist Rate and Nyquist Interval Nyquist rate : When the sampling rate becomes exactly equal to '2W" samples sec, for a given bandwidth of W Hertz, then it is called Nyquist rate. Nyquist interval : It is the time interval between any two adjacent samples when sampling rate is Nyquist rate. Nyquist rate 2W Hz = 13.7) 1 say Seconds + (1.3.8) Nyquist interval = 7, 4.3.5 Reconstruction Filter (Interpolation Filter) Definition In section 13.2 we have shown that the reconstructed signal is the succession of sine pulses weighted by x(nT,). These pulses are interpolated with the help of a lowpass filter. It is also called reconstruction filter or interpolation filter.Digital Communications 1-18 Pulse Digital Modulation Ideal filter Fig. 13.7 shows the spectrum of sampled signal and frequency response of required filter. When the sampling frequency is exactly 2W, then the spectrums just touch each other as shown in Fig. 13.7. The spectrum of original signal, X(/) can be filtered by an ideal filter having passband from -W 0, ~ Op 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 1-24 Pulse Digital Modulation *A fimelimited signal which is zero for [Hs igi termine By Merle of its frequency spectrum at intervals less than + Hertz apart". «Explanation : Thus the spectrum is sampled at f, <7 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 cignal. 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 1311 (@) that a rectangular pulse is time limited to +2 seconds i, xO=A for -Est Shah oa |) eae 2W )@xWe= nn) = yf.) sin QWE-n) . Ez) me We=ny above equation becomes, Since sinc 0 =: as jsinc(2We-n) -e o v bx 09 ‘al - Ly | Fig. 1.4.7 (a) Basoband signal x (t) | {b) Instantaneously sampled signal x, (¢) | (c) Constant pulse width function A(t) | {@) Flat top sampled signal s(t) obtained through convolution of A (t)and x; (t)Digital Communicati Pulse Digital Modulation J £O8@-%) = feo) (1.4.16) Using this equation we can write equation 1.4.15 as, st = Sxintnt-n7,) (1417) * This equation represents value of s(t) in terms of sampled value x(n T,) and function h (t~ T,) for flat top sampled signal. we also know from equation 1.4.12 that, s®) = x 0°hO By taking Fourier transform of both sides of above equation, SP) = XPH(N vs (1.4.18) Convolution in time domain is converted to multiplication in frequency domain. Xs (9) is given as, XM =£ EXU-nf) w= (1.419) : Equation 1.4.18 becomes, Spectrum of Hat Top Sampled Signal : S(=f, 5 XUF-mf HI) | 420) This equation represents the spectrum of flat top sampled signal. 4.4.3.4 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(f) is obtained by passing through a filter having transfer function H (f). The corresponding impulse response /(0), in time domain is shown in Fig, 1.48 (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 14.20 represents that spectrum of this rectangular pulse is multiplied with that of x(t) Fig. 1.4.8 (b) shows the spectrum of one rectangular pulse of 1(t) The spectrum of a rectangular pulse is given as, H() = tsine(f yeni sA=1 w= (1421)Digital Communications 141 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() from s(t). The high frequency rollotf 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 ‘v’.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. Message ‘signal x) Fig. 1.4.9 Recovering x(t) From equation 14.21 we know that the sample function f(t) acts like a lowpass filter where Fourier transform is given as, HP = tsinc(ft)e7#* from equation 14.21 (14.22)Digital Communication: This spectrum is plotted in Fig, 148. Equalizer used in cascade with the recoristriiction 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, Ker Penta Ha = FG wa (1.4.23) Here ‘ty’ is the delay introduced by lowpass filter which is equal to t/2 Kei He(f = —~£—__ ah tsine (fee kK * ysinc() sw (14.24) ‘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 ct | It uses multiplication by | It uses chopping It uses sample and ‘sampling ‘an imputse function | principle hold creuit 2 | Circuit of sampler Samping Discharge $ eto AO. x(t) i xt) Gy Cth i ; x) sth 3 | Wavetorns x x0 7) xg) x04) (ot) t a = T]he 7 4 | Resteabitty “hist not practically | This mothod is weed possible methed | practicallyDigital Communications 4-43 Pulse Digital Motiulation Samping rata | Samping rato tenes | Sanping rate satstes | Samping rte eases ‘ern Tyee esta | Nyael, eters 6 | Noise Nicise inerference t= | Noise erference is | Nowe intererence fs bterference rmsimure ‘nasimum 7 [rime a = = sentaton | so-7 E eae ¥ x(0T,)5(t-0Ts) x (she (nf, 2) x(nT,)h-nT,) oie! 3 | Fr = = > éonain xt E sof smn d representation o x-ms) sme nnoxe—ny | x-atane Table 1.4.1 Comparison of sampling techniques tmp Example 1.4.1: The spectrum of signal x(0) shown below. This signal is sampled at the Nyquist 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. 4 = 15 7 Fig. 1.4.10 Solution : It is clear from Fig, 1.4.10 that the signal is bandlimited to 10 Hz. W = 10H Nyquist rate = 2xW=2x10=20Hz Since the signal is sampled at Nyquist rate, the sampling frequency will be, f = Hz 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, 8 = & DXU-nfHO 145)Digital Gommunications 4 Pulse Digital Modulation ‘Value of H(f) is given by equation 1.4.21 as, H(f) = tsine(fr) “8st ow (1.4.26) Here + is the width of the rectangular pulse used for sampling. The given value of rectangular sampling pulse is 50/3 milliseconds. ic., ts Bacio or + = 28 seconds Putting the value of ¢ in equation 1.4.26 we get, AY) = SP ne 99 Jerome Put this value of H () and fin equation 14.25 sip = 20% p20 02 sine Ee sooner (since f, = 20) sp = 3 x20 esr SSE fo097> 3 This expression gives the spectrum up to 60 Hz (since n=3) for the sampled signal. im Example 1.4.2 : A flat top sampling system samples a signal of maximum 1 Hz with 25 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; = 2.5 Hz Maximum signal frequency fy, = 1 Hz Pulse width += 02 sec. By equation 1.422, the aperture effect is given by a transfer function H(f) as, H(f) = tsine(fye“ isDigital Communications 1-45 Pulse Digital Modulation ‘The magnitude of the above equation is given as, JH(D| = tsinc(t) .. (14.27) JH (P| = O2sinc(fx 02) Aperture effect at highest frequency will be obtained by putting f= finax 1 Hz in above equation ie, 1H()| = 0.2 sine (0.2) = 0.18709 or IH@| = 18.70% w= (Ans) From equation 1.424 the equalizer characteristic is given as, k HeA = Tanege Putting 1=02 second and assuming k=1, the above equation will be, 1 Ha * SEmmeOEp w» (1.4.28) This equation is the plot of H,,(/)Vsf and it represents the equalization characteristic to overcome aperture effect. 1.4.5 Transmission Bandwidth of PAM Signal The pulse duration ‘ris 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, f = Wor 1 1s T. $ apy since f= 1 and ti Rsay vn (1.4.29) If ON and OFF time of the pulse is same, then frequency of the PAM pulse becomes, feted (4.4.30) Thus Fig. 14.11 shows that if ON and OFF times of PAM signal are Fig. 1.411 Maximum frequency of PAM samme, then maximum frequency of signi t 7Digital Communications 1-46 Pulse Digital Modulation i PAM signal is given by equation 1.430 ie,, Foo = z ~ (1431) ‘« Bandwidth required for transmission of PAM signal will be equal to maximum frequency fr given by above equation. This bandwidth gives adequate pulse resolution ie,, Br 2 frrax Bree = (1.4.32) Since v>W w= (14.33) Thus the transmission bandwidth By of PAM signal is very very large compared to highest frequency in the signal (1). 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 diagrana explain the working of a circuit that provides flat top sampling. 2. Show that « 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 samplmg_(b) Flat top sampling ? 3. What is aperture efect ? How it can be reduced ?Digital Communications 1-47 Pulse Digital Madulation 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 amplitide 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. Time (a) Flat Top PAM Lim | i] SS ML. 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. 1.52 (a) shows the scheme to generate PDM and PPM. The comesponding waveforms are shown in Fig. 15.2 (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 x) @ xt) Time A PDM te sy Time Kr A PPM Time We 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(t) 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(f). 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(t). If the sawtooth waveform is reversed, then trailing edge will be fixed and leading edge will beDigital Communications 1-49 Puise 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(l) 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., << T; And transmission bandwidth should be, 1 Br z oa ‘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 : Br ae ~ (1.51) 1.5.3 Comparison between Various Pulse Modulation Methods Following table shows the comparison among various pulse modulation techniques.Digital Communications 1-50 Pulse Digital Modulation 7 Sr. |Puise Amplitude Modulation] Pulse Width/Duration Pulse Fosition Modulation No. ‘Modulation 1 Wavelorm Waveform Waveform Time Time 2 | Amplitude of the pulse is Width of the pulse is ‘The relative position of the proportional to amplitude of | proportional to amplitude of pulse Is proprotional to the modulating signal. modulating signal, amplitude of modulating signal. 3] The bandwidth of the Bandwidth of transmission Bandwidth of transmission tranemissien channal depends! channel depends on rise time of| channel deponds on rising ‘on width of the pulse. the pulse. time of the pulse. 4 | The instantaneous power of | The instantaneous power of the | The instantaneous power of the trarsmitter v transmitter varies. the transmitter remains constant. 5 | Noiso interfarance is high. | Noise interference is minimum. | Noise interforen ‘rinimum. 6 | system is complex. ‘Simple to implement. ‘Simple to mplemert, 7 amplitude ‘Similar to frequency modulation, | Similar to phase modulatior = Table 1.5.1 Comparison of PAM, PPM and PDM im Example 1.5.1: For a PAM transmission of voice signal with W = 3 kHz. Calculate Br if f, = 8kHz and t=01T,. Solution : ts of 1 T, is given as, 7, = 7=—1 8 given ax103 2 <= 017, =O, sec 8x105 From equation 1.5.1, the transmission bandwidth By is given as, 1 1 By = 3 2——__ = 40 kHz "> 8x103Digital Communications 4-54 Pulse Digital Modulation ‘a> 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 minima transmission bandzoidth needed for PDM and PPM. Solution : In example 15.1 we obtained the pulse width t= =a see, The rise time is given as 1% of width of pulse ic., fy = rx00=—, xa01 = 125x107 see 3x10 We know that transmission bandwidth is given as, 1 ———, 24 MHz 2x 1.25107 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 : 3) For both systems, the figure of merit is proportional to square of the ratio Br wl ii) As the signal to noise ratio is reduced, both the systems exhibit threshold effect. + With digital pulse modulation, the beiter 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 (PAM/TDM 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. 4.74 Block Diagram of PAM / TDM Fig.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 J. The single signal composed due to multiplexing of input channels is given to the transmission channel. At the receiver the decommutater separates (decodes) the time multiplexed input channels. ‘These channel! signals are then passed through lowpass reconstruction filters. e Fig, 1.7.1 TDM system (PAMITDM system) (a) Block diagram —(b) WaveformsDigital Gommunications 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, i 2 W (174) ‘Therefore the time space between successive samples from any one input will be T= r wn (172) 1 Sw (173) ‘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 equal to, os Spacing between two samples = ¥ (174) pn? channel pulse (n4) channel pulse TyN TN Fig. 1.7.2 Calculation of number of pulses per second in TOM From the above figure we can very easily calculate the number of pulses per second or pulse frequency as, Number of pulses per second= Spacing betweenDigital Communications 1°54 Pulse Digital Modulation ‘We know that T, = N 17h These number of pulses per second is also called signalling rate of TDM signal and is denewd by 'T ie., +. Number of pulses per second = Y= N f, (175) Signalling rate = r=N f, +» (17.6) ce fi > 2W, then signaling rate becomes, a aes I Signalling rate in PAM/TDM system : r > 2NW (177) ULC EEE Eee ‘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. ie, 14 By = Sr=sNh ara (78) *. Transmission bandwidth of TDM channel will be equal to bandwidth of the lowpass filter, : By = BN. fk from above equation Hf sampling rate becomes equal to Nyquist rate i., f.(min) = Nyquist rate = 2W, then by = Lweaw Minimum transmission bandwidth of TDM channel : By = NW os (179) This equation shows that if there are total ‘N’ channels in TDM which are bandiimited to "W" Hz, then minimum bandwidth of the transmission channel will be equal to NW.Digital Communications 1-55 Pulse Digital Modulation mp Example 4.7.4: '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. Impulses from various channel | {one framo) 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 ‘7,’ 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 = # wn (17.10) Since maximum signal frequency is fy, the minimum sampling frequency will be f.=2%y (ie. minimum sampling rate or Nyquist rate). 1 1 rede SA om ‘Therefore equation 1.7.10 will be, Spacing between two consecutive samples = RE ww 17.11) ‘The impulse train of Fig, 1.7.3 is given to PAM/TDM transmission channel. This channel is lowpass type of channel as shown in Fig. 1.74.Digital Communications 1°56 Pulse Digital Modulation Output x) Le ‘impulses from varous channels Lowpass type tansmission channel Fig. 1.7.4 PAM/TDM transmission channol ‘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(/)|=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(t). Therefore, Impulse response of the transmission channel = /(f)= IFT (H (/)] Since output y(t) is nothing but impulse response of transmission channel (since input x(1) is train of impulses), y@® = hO=ETIH() = fers af By definition of IFT. h = freien ap Since H(f)=1for-f, $f Sf; he _ fetat Ye _ elt — ex Pate * Gar |, ar 1 [eit — fae a o a sin 2nf..) [By Euler's theorem] (1.7.12) = yf, Bor By rearranging the equation = 2f, sinc(2f.0 ~+- (1.7.13)Digital Communications 1-57 Pulse Digital Modulation ‘Thus the output is a sinc function and we know that it has zero values when 2fet = £1,42,43,24, . A 142,354 ie. te ttt te, Be" Be fe” fe This can also be verified from equation 1.7.12, At above given values of t, sin (2nf(4) has zero values, Fig. 1.7.5 shows the plot of sinc function. 0 ‘The amplitudes of sine pulses eo are weighed by the amplitudes of thelr impulses. 2g Xe \ Responses due to \ various impulses go | \ to zero at these points duetox, is —dueto x, i “F088 talk applied 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 t=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 response due to this impulse also persists over long period. This means at any time the responses due to other impulses are present. Therefore there is possibility of crosstalk. But a careful observation of Fig. 1.7.5 shows that responses due to all the ‘i att +2 ,+ = other impulses are at t= 457,255 /t5¢ tgp, except that of impulse sent at that time. For example at t=0, responses due to all other impulses are zero exceptimpulse response due to x, it has peak value of =0. Similarly at response due to x is at peak whereas all other responses are zero. This shows that if impulses are transmitted at =0, tae the crosstalk will be zero. In other words we can say that the spacing between two consecutive samples should be 1, i : Fp B order to avoid crosstalk be, spacing between two consecutive samples in order to avoid crosstalk = 37- (1.7.14) Comparing the above equation with equation 1.7.11 (which also gives spacing between two consecutive samples), i=! fe 2N fa fe = Nfn Thus, Minimum channel bandwidth to avoid crosstalk : f.=N fin (17.15) Observe that this equation is similar to the relation we obtained earlier given by equation 1.79. 1.1.2 Synchronization in TOM 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. 17.6 shows the TDM signals with markers. Marker 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 (ie, baseband signal) by passing through a baseband filter. Fig. 1.77 shows the TDM transmission with baseband filtering and the baseband waveform. on Scorn" |__| | ol rrodiator a x . to] Moca a © Fig. 1.7.7 (a) TDM 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 crossialk 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 Ty 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, Ag = Act .- (1.7.16)Digital Communications 1-60 Pulse Digital Modulation A This decay gives isa to erosetalk K cuaainw 78 (a) A rectangular pulse applied to the lowpass channol {8} Respones of the lowpass chonnel to the rectangular pulse And the cress talk reduction factor is defined as, Ay Ky = r0ter( 4) = -S4SBT, dB (7.47) This equation shows that to keep cross talk below -30dB ,T, should be greater than <4, The guard times are very much important particularly in pulse duration or pulse position modulation techniques. ‘mp Example 4.7.2 : 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 = Nfn (By equation 1.7.15) 12« 10 kHz = 120 kHz im 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 modulaied 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 15x 10° bits/sec. Solution : i) We know that if N channels are time division multiplexed, then minimum transmission bandwidth is given as, By = NW Here W is the maximum frequency in the signals. By = 24x34kHz=816kHz ii) The signalling rate of the system is given as, 1 = 1.5x10° bits/sec Since there are 24 channels, the bit rate of an individual channel is, 1,5x10° + (one channel) a 62500 bits/sec Since each sample is encoded using 8 bits, the samples per second will be, F (one channel) bits/sec Sample/see = ee ‘Samples per seconds is nothing but sampling frequency. 62500 bits/ see Bbits/ sample f= = 78125 Hz or samples per second mip Example 17. . (Ans) . (Ans) + Twenty four voice signals are sampled uniformly and then time division multiplexed. The sampling operation uses flat samples with 1 usec duration. The multiplexing operation provides for synckronization by adding an extra pulse of Iusec 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, =i. 1. i ~ 8000 = 125 psec T,Digital Communications 4-62 Pulse Digital Modulation Thus in 125 psec time there are 25 pulses at uniform distances. This is illustrated in Fig. 1.79. Fig. 1.7.9 Multiplexing of 24 voice signals As shown in above figure, the pulses are separated by oe = 5s. Width of the pulse is 1 ys. Hence, Spacing between pulses = 5-1 = 4 psec. Fig. 1.7.10 shows the multiplexing scheme. — x vice J % lutiplexee Muttiplexed one Mai oa wet stor oe BB sans Fig. 1.7.10 PAM-TDM system Theory Questions 1. Explain PAM/TDM system for 'N’ number of channels, 2. Derive the relation for minimum bandwidth to transmit “N° channels in PAM/IDM system such thal crosstalk is avoided. 3. Explain the importance of synchronization in TDM systems.Digital Communications 1-63 Pulse Unsolved Examples 1. Twenty four voice signals ere sampled uniformly and then time division multiplexed, the saripling operation uses flat top semples with 1 ysec duration. The synchronization is provided by adding an extra pulse of 1 yssec duration. The highest frequency component of each voice signal is 3.4 kHz. (a) For sampling rate of 8 Fz, calculate spacing between successive pulses of multiplexed signal () 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(t) at frequency f, 22W. This sampled ‘Variable amplitude’ pulse is then digitized by the analog to digital converter. The parallel bits obtained are converted to a serial bit stream. Fig.18.1 shows the PCM generator. vdigis Lowpass (nf.)| Binary Parallel PCM x0 iter Pe encase toaeral -—> few (digitizer) convener | (=v%, {220 Fig, 1.8.1 PCM generator In the PCM generator of above figure, the signal x(t) is first passed through the lowpass filter of cuto!f 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 fz. Sampling frequency f, is selected sufficiently above Nyquist rate to avoid aliasing ie, f= Ww In Fig. 18.1 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 T,) with its fixed digital levels. It then assigns any one of the digital level to x(n T,) which results in minimum distortion or error. This error is called quantization error. Thus output of quantizer is a digital level called 3, (nT)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, (17,) is converted to 'V' binary bits. The encoder is also called digitizer. It is not possible to transmit each bit of the binary word separately on transmission line. Therefore '#) 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 ‘o’ number of binary digits to represent each level. Then the number of levels that can be represented by 'o’ digits will be, qeP (181) Here ‘g' represents total number of digital levels of g-level quantizer. For example if v=3 bits, then total number of levels will be, 4 = 23 =8 levels Each sample is converted to 'v' binary bits, ie. Number of bits per sample = ‘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) = » bits por sample x, samples per second... (1.82) ‘The number of bits per second is also called signaling rate of PCM and is denoted by 'rie, Signaling rate in PCM: r = vf, +» (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., 1 Br 25r ~~ (1.84) ‘Transmission Bandwidth of PCM: {Br = 3 vf, Since f, 22W += (18.5) Br 20W - (1.86) 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. vdigts PCM+ Noise Sia 3 ° ) 14 114 Fig. 1.8.2 (a) POM 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 ypy(H} As shown in reconstructed signal of Fig. 1.82 (b), it is impossible to reconstruct exact original signal x(f) 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 bits 'v' increases the signaling rate as well as transmission bandwidth as we have seen in equation 1.83 and equation 1.86. ‘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. 1.8.3. As shown in this figure, when an input is between ~ 6/2 and + 3/2 then the quantizer output is zero. i.e., For ~8/2 § x(nT,) < 8/2; “xq (nT) = 0 Here is the step size of the quantizer. for 8/2 $ x (nT) < 38/2; x (nT) =5 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 (aT) -x (nT) a= (18.7) In Fig. 18.3 (b) observe that when x(nT,) = 0, x,(nT,) = 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. 18.3 (b) it is clear that, ~8/2 < es6/2 so» (1.88) ‘Thus quantization error lies between - 8/2 and + 8/2. And maximum quantization 3 error is, maximum quantization error, € max -= (189)Digital Communications 1-67 Pulse Digital Modulation “Quantizer output | Ideal tanster characteristic passes through zero Staircase approximation ty [re | Fig. 1.8.3 (a) Quantization characteristic of midtread quantizer (b) Quantization error 1.84.2 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 5, the output is 5/2. Similarly when an input is between 0 and - 8, the output is ~ 8/2. ie., For 0 < x(nl,)<8; 4 (aT) = 5/2 ~8s x (nl) <0; x (nT) = - 3/2 Similarly when an input is between 3 8 and 4 5, the output is 7 8/2. This is called midriser quantizer because its output is either + 6/2 or ~ 6/2 when input is zeroDigital Communications 1-68 Pulse Digital Modulation i Peer t+ Cry rr CTT I Fig. 1.8.4 (a) Transfer characteristic of midriser quantizer (b) Quantization error Fig. 184 (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 (at) = 6/2-0=8/2 ‘Thus the quantization error lies between — 6/2 and + 8/2. ie., -8/2 s es 8/2 = (1.8.10) And the maximum quantization error is, = i3| os (18.11) EmaxDigital Communications 1 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. 1.8.43 Biased Quantizer Fig. 1.8.5 shows the transfer characteristic of biased uniform quantizer. LT] Guantzeroupur [. x40) ++ Prrry Fig. 1.8.5 (2) 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. i.e, for 0S x (nT) <3; xq (nT) ~0Digital Communications 1-70 Pulse Digital Modulation Similarly, for -8 S x (nT,)< Oj x (nT) =-8 Fig. 185 shows quantization error. When input is 8 output is zero. Hence quantization error is, € = xq (al) - xin) = 0-8=-8 ‘Thus the quantization error lies between 0 and - 3. ie., -8 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 « uniform quantizer followed by a v bit encoder. Show that rins signal to quantization noise ratio is approximately given by (18 + 60) 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, vi Pee Here V = rms value [An 27° vn (1839) When R =1, the power P is normalized, ie, Nommalized power : P 4 with R =1 in above equation. + Signal to quantization noise ratio is given by equation 18.33 as, SP & x22 NO xhax Here PDigital Communications 1-76 Pulse Digital Modulation Expressing signal to noise power ratio in dB, Ss s (x) = 10log 40 (5)- 10 log jg (1.5%2™) 10 log 9 (1.5) +10 log 19 2 = 1.76+20x10%x03 Thus, S Vipin rom :( 5 ap = 1.8-+60 ; for sinusoidal signal 7% }eBin roo :( § lap = 1.860; for sinusoidal signal -. (1.8.40) mb Example 1.8.2: A Television signal with a bandwidth of 4.2 MHz is transmitted using binary PCM. The muriber of quantization levels is 512. ‘Calculate, i) Code word length ii) Transmission bandwidth iti) Final bit rate ie) Output signal to quantization noise ratio. (March-2003, 10 Marks] Solution : The bandwidth is 42 MHz, means highest frequency component will have frequency of 42 MHz ie, W = 42 MHz Quantization levels ¢ = 512 i) Number of bits and quantization levels are related in binary PCM as, gee ie. siz = 2" log 512 = vlog2 or y = 1g512 Tog2 = 9 bits w= (Ans) ‘Thus the code word length is 9 bits. ii) From equation 1.86 the transmission channel bandwidth is given as, B, > ww 2 9x42%109 Hz By = 378MHz w+ (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 = 84 MHz Putting this value of 'f,’ in equation for signaling rate, 9x84 x106 = 756x10° bits/sec s+ (Ans) r From equation 1.84 transmission bandwidth is also obtained as, 1 Br 2 37 4756x106 bits/sec wv or By 2 378 MHz which is same as the value obtained earlier. iv) The signal to noise ratio s (a)# < 48+60dB v S 484+6x9 $988 dB s+ (Ans) imap Example 1.8.3 : The bandwidth of signal input to the PCM is restricted to 4 kHz. The input varies from ~38 V to + 38 V and has the average power of 30 mW. The repuired signal to noise ratio is 20 dB. The modulator produces binary output. Assume uniform quantization. i) Calculate the number of bits required per sample. ii) Outputs of 30 such PCM coders are time multiplexed. What is the minimum required transmission bandwidth for the multiplexed signal ? Solution : ‘The given value of signal to noise ratio is 20 dB. t Ss Ss ie ($}e = wre (§)-200 Si. = 100Digital Communications 1-78 Pulse Digital Modulation i) The signal to quantization noise ratio is given as, Ss 3p.2% . Nt ye By equation 1.833 Here Xwmax = 38V.P= 30mW and © = 100 530%10-3 22 L too = 3%30%103 2 Ga b P= 698 bits = 7 bits ++ (Ans) ii) The maximum frequency is, W = 4 kHz ‘The transmission bandwidth is given by equation 1.8.6 as, By 2 ow Since there are 30 PCM coders which are time multiplexed, the transmission bandwidth will be, By = 30x07 W > 307 x4 KH: = 840 kHz s+ (Ans) Signaling rate is two times the transmission bandwidth as given by equation 1.8.4 ie, Signaling rate r = 840x2 bits/sec = 1680 bits/sec. wm 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 +01%. That is quantization error should be £01%. or the maximum quantization error should be +01% £01% or e, £0001 ‘The maximum quantization error for an uniform quantizer is given as, cm « B or 0.001 ‘That is 2x0.001 = 0.002 Step size 6 The step size, number of levels and maximum value of the signal are related as (By equation 18.16) or = 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, f2w 2 21002200 Hz w+ (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, g=P Here, 4g = number of levelsDigital Communications 4-81 Pulse Digital Modulation ‘Thus the maximum message bandwidth is 357 MHz. b) The modulating wave is sinusoidal. For such signal, the signal to quantization noise ratio is given by (3) dB i> Example 1.8.6 : fu= = 18+6v By equation 1.840 = 18+6x7 (putting for v=7) = 438 dB w (Ans) The information in an analog waveform with maximum frequency KEz is to be transmitted over an M-level PCM system where the numter of pulse levels is M=16 The quantization distortion is specified not to exceed 1% of peak to peak analog signal. i) What is the maximum nuraber of bits per sample that should be used in this PCM system ? if) What is the minimum sampling rate and what is the resulting bit transmission rate? Solution ; i) Since the number of levels given here are M=16, q = M=16 ‘Then bits and levels in binary PCM are related as, q 16 or v il) Since Sua minimum f, “ f or =P = bits 22 24 W=3kHz 0 v 2w by sampling theorem > 2x3kHz > 6kHz vue (Ans) Bit transmission rate or signaling rate is given by equation 1.8.3 as, r = of 4x6%103 w v 24x103 bits per second a (Ans)Digital Communications 4-82 Pulse Digital Modulation im Example 1.8.7 : A Signal of bandwidth 3.5 kHz is sampled quantized and coded by a PCM system. The coded signal is then transmitted over a transmission channel of supporting a transmission rate of 50 k bits/sec. Caiculate te maximum signal to noise ratio that can be obtained by this system. The input signal has peak to peak value of 4 volts and rms value of 0.2 V. Solution : The maximum frequency of the signal is 35 kHz, ie. W = 3.5kHz ‘Therefore sampling frequency will be f 2 Ww 2 2x3.5kHz 2 7 kHz ‘The signaling rate is given by equation 1.83 as, r=ovf Putting values of r =50x10? bits/sec and f, 27x10° Hz in above equation. 50x10? < v-7x105 o v 2 7.142 bits v = 8 bits The rms value of the signal is 0.2 V. Therefore the normalized signal power will be, (027 oT ie, P= OW Normalized signal power [IR = 1 for normalized power] The maximum signal to noise ratio is given by S _ 3P.2 N x, 2 in above equation, S _ 3x004x2?*8 N @ Putting the values of P: = 196608 ~ 33 dB wm Example 1.8.8 : A signal x({) is uniformly distributed in the range + Xjyqq. Calculate maximum signal to noise ratio for this signal.Digital Communications 1-83 Pulse Digital Modulation Solution : he signal is uniformly distributed in the range +x,,,,- Therefore we can ‘write its PDF (using the Standard Uniform Distribution) as, KO fl 0 for X< my 1 2X max £0 —Zmay © X Xmmax Fig. 1.8.7 shows this PDF, 10) & 1 1 a art Fae) Fig, 1.8.7 PDF of a uniformly distributed random variable ‘The mean square value of a random variable X is given as, FO= P20 pw . 1 feop™ ' © Fimo [FL The signal power P zo Normalized signal power P = = [since R=1]Digital Communications 1-84 Pulse Digital Modulation Xoax “3 2x, Step size b= Sime By equation 18.16 = 4 Xm = 8242 s-Normalized signal power, P = 5 z 2 Nonmalized noise power = Ea By equation 1.8.29 “Sj . fo §. = Normalized signal power «Signal to noise power ratio 5 = Sect pee power 82g? /12 ey Since q=2°, above equation will be, S 2 op no? or Gi \s = log yg (2) aB ~ 60 This is the required expression for maximum value of signal to noise ratio. wm Example 1.8.9 : Consider an audio signal comprised of the sinusoidal term 3 (0) =3.c0s (600K4) i) Find the signal to quantization noise ratio when this is quantized using 10 bit PCM. ii) How many bits of quantization are needed to achieve a signal to quantization noise ratio of alleast 40 dB? Solution : Here s(() = 3.cos (500 xt) That is sinusoidal signal applied to the quantizer. i) Let us assume that peak value of cosine wave defined by s(t) covers the complete range of quantizer. ie. An, = 3V covers complete range.Digital Communications 1-85 Pulse Digital Modulation We know that signal to noise ratio for sinusoidal signal is given by $ (5 \e = 18+60 Here 10 bit PCM is used ie, eo = 10 (6 ii) For sinusoidal signal again we will use the same relation. ie. ) Jie = 18+ 60dn 1.8 dB 18+6x10 ie. ( 5 To get signal to noise ratio of at least 40 dB we can write above equation as, 18+6v 2 40dB v 2 6.36 bits = 7 bits Thus at least 7 bils are required to get signal to noise ratio of 40 dB. om Example 4.8.10: A 7 bit PCM sysiem employing uniform quantization has an overall signaling rate of 56 k bits per second. Calculate the signal 10 quantization noise ratio that would result when its input is a sine wave with peak to peck 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 2 = 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, (s} By equation 1.83 signaling rate is given as, re of 18+ 60dl =18+6x7 438 dBDigital Communications 1-86 Pulse Digital Modulation Putting 1 =56x 10° bits/second and v=7 bits in above equation we get, 536x103 = 7-f, -Sampling frequency, f, = 8x103 Hz By sampling theorem, f, 2 2W Maximum frequency that can be handled is given as, fi < 8000 wsos W s 4000 Hz (Ans) wm 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 i 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 4 = 1024 levels ‘The Nyquist rate is, 2We Nyquist rate x45 = 9 MHz ‘The sampling rate is 20% above the nyquist rate. ie. Sampling rate, f, = 1.2 x9 = 108 MHz We know that quantization levels q and number of bits v are related as, q= 2 024 = 2” o 2 = 10bits ‘The number of bits/sec generated by PCM system is called bit rate or signaling rate. ie, Signaling rate, r " vf, 10 x 108 x 10° bits/sec. 108 x10 bits / sec. 0Digital Communications 1-87 Pulse Digital Modulation ‘The output bit rate does not change if linear PCM is converted into companded PCM. Companded PCM is used to improve the signal to noise ratio. 1.8.6 Nonuniform Quantization In nonuniform quantization, the step size is not fixed. It varies according to certain law or as per input signal amplitude. Fig. 1.8.8 shows the transfer characteristic and error in nonuniform quantization. TEA 3a Zl ee Sea Cee et t t Fig. 1.8.8 (a) Nonuniform quantization transfer characteristic (b) Quantization error In this figure observe that step size is small at low input signal levels. Hence quantization error is also small at these inputs. Therefore signal to quantization noise power ratio is improved at low signal levels. Stepsize is higher at high input levels. Hence signal to noise power ratio remains almost same throughout the dynamic range of quantizer.Digital Communications 1-88 Pulse Digital Modulation 1.8.6.1 Necessity of Nonuniform Quantization In uniform quantization, the quantizer has a linear characteristics as we have seen in Fig. 1.84 (a). The step size also remains same throughout the range of quantizer. Therefore over the complete range of inputs, the maximum quantization error also remains same, From equation 1.8.11 the quantization error is given as, ‘Maximum quantization error = - (1.8.41) From equation 1.8.16 step size'S'is given as, 2xmae q é= If x(0 is normalized, its maximum value i. Xyyay =1. (1.8.42) Let us consider an example of PCM system in which ‘Thea number of levels q will be, q = 4 =16levels, <. From equation 1.8.42 the step size 8 will be, = 222.1 “G68 --Quantization error is given from equation 1.8.41 as, . § cm =f ria 2x8, 16 ‘Thus the quantization error z volts of the full range voltage. For simplicity, assume that full range voltage is 16 volls. Then maximum quantization error will be 1 volt. For the low signal amplitudes like 2 volts, 3 volts etc, the maximum quantization error of 1 volt is quite high ie. about 30 to 50%. But for signal amplitudes near 15 volts, 16 volts etc., the maximum quantization error (which is same throughout the range) of 1 volt can be considered to be small. This problem arises because of uniform quantization. Therefore nonuniform quantization should be used in such cases, Another example is discussed next.Digital Communications 4-89 Pulse Digital Modulation 1.8.6.2 Nocessity of Nonuniform Quantization for Spooch Signal We know that speech and music signals are characterized by large crest factor. That is for such signals the ratio of peak to rms value is very high. Crest factor = ees (1.8.43) = Very high for speech and music. We know that the signal to noise ratio is given as, S = (3x2%P) By equation 1.8.35 (1.8.44) Expressing in decibels, (a) = logy (3x22 xP) ) If we normalize the signal power ie. if P =1, then above equation becomes, (5) = (48460) dB wes (1.8.45) Here power P is defined as, P= st! 2 0. Pp P= EN V3.,q = Mean square value of signal voltage = 20 i! 20 i ©. Normalized power will be, p= <2 with R=7] Pax (1.8.46) Crest factor is given as, | __ Peak value Xmay Crest factor = Eis * aa .. (L847) x ) = Xmx since P = x? () (1.8.48) | AP | When we normalize the signal x(0), then i Xmax = 1 .. (1.8.49) Putting above value of xp), in equation 1.8.48, Crest factor = —L (1.8.50) vPDigital Communications. 1-90 Pulse Digital Modulation For a large crest factor of voice (speech) and music signals P should be very very less than one in above equation. ie, Picct for large crest factor in equation 1.8.50 Therefore actual signal to noise ratio will be significantly less than the value that is given by equation 1.8.45, since in this equation P=1. Consider equation 18.44, (3) = 3x2" xP (42x Ipcey << OK" 451) ‘This equation shows that the signal to noise ratio for large crest factor signal (P<<1) will be very very less than that of the calculated theoretical value. The theoretical value is obtained for normalized power (P=1) by equation 1.8.44. Therefore such large crest factor signals (speech and music) should use nonuniform quantization to overcome the problem just discussed. Signal to noise ratio reduces at low power levels (P-<<1) just now we have seen by equation 1.8.51. That is at low signal levels, signal to noise ratio reduces mean noise increases. The quantization noise is directly related to step size. Therefore at low signal levels (P<<1) noise can be kept low by keeping step size low. This means that at low signal levels signal to noise ratio can be increased by decreasing step size ‘8. This means step size "8' should be varied according to the signal level to keep signal to noise ratio at the required value. This is nothing but nonuniform quantization. Now let's see how nonuniform quantization is achieved through companding, 1.86.3 Companding in PCM Normally we don’t know how the signal level will vary in advance. Therefore the nonuniform quantization (variable step size '8) becomes difficult to implement. Therefore the signal is amplified at low signal levels and attenuated at high signal levels. After this process, uniform quantization is used. This is equivalent to more step size at low signal levels and small step size at high signal levels. At the receiver a reverse process is done. That is signal is attenuated at low signal levels and amplified at high signal levels to get original signal. Thus the compression of signal at transmitter and expansion at receiver is called combinely as companding. Fig. 1.8.9 shows compression and expansion curves. As can be seen from Fig. 1.8.9, at the receiver, the signal is expanded exactly opposite to compression curve at transmitter to get original signal. A dotted line in the Fig. 1.89 shows uniform quantization. The compression and expansion is obtained by passing the signal through the amplifier having nonlinear transfer characteristic asDigital Communications 1-91 Pulse Digital Modulation shown in Fig. 18.9. That is nonlinear transfer characteristic means compression and expansion curves. ou Compression Expansion Linear charactaristies Compression at transmiter Fig. 1.8.9 Companding curves for PCM 1.864 y+ Law Companding for Speech Signals Normally for speech and music signals a ji - law compression is used. This compression is defined by the following equation, In +ula) 20) = SOT ep) |x}st ++ (1.8.52) Fig. 18.10 shows the variation of signal to noise ratio with respect to signal level without companding and with companding. Without companding 49-30 20 “to 0 Signal level 4B —> Fig, 1.8.10 PCM performance with 1 - law compandingDigital Communications 1-92 Pulse Digital Modulation It can be observed from above figure that signal to noise ratio of PCM remains almost constant with companding. 1.88.5 ALaw for Companding The A law provides piecewise compressor characteristic. It has linear segment for low level inputs and logarithmic segment for high level inputs. It is defined as, Alsl for os|x|s2. zy = 4, bein 4 (1.8.53) © /AemAlD fy Letsiet on (18 1+hA A When A = 1, we get uniform quantization. The practical value for A is 87.56. Both A-law and jtlaw companding is used for PCM telephone systems, 4.86.6 Signal to Noise Ratio of Companded PCM ‘The signal to noise ratio of companded PCM is given as, -_3r iimare o> (1.8.54) Here q = 2° is number of quantization levels. ium Example 1.8.12; For a random variable are the mean square value and variance akoays equal ? Calculate these quantities for the quantization noise or error in PCM system. Solution : For a random variable X, the variance G? is given as, 02 = X?-m? Here X? is the mean square value and m, is the mean value. Above equation shows that variance (0?) and mean square value (X?) will be same if mean (m,) is zero. From quantization characteristics of Fig. 1.84 (b) it is clear that quantization error (e) has zero mean or average value. And it follows uniform distribution from ~® to +2. Hence probability density function of quantization error will be,Digital Communications 4-93 Pulse Digital Modulation 1 fF.) = 43 By equation 1.8.22 0 elsewhere Mean square value can be calculated as, XP = [xp ar Putting values in above equation, a? ; 3 _ fe? 2 = ls[e 3 5 “2 ‘This is the mean square value of quantization error. im Example 1.8.13 : A compact disc (CD) records audio signals digitally by using PCM. Assume the audio signal bandwidth to be 15 kHz. (i) What is Nyquist rate ? (id ‘If the Nyquist samples are quantized into L = 65536 levels and then binary eoded determine the nuntber of binary digits required to encode a sample. ii) Determine the number of binary digits per second (bits/sec) required to encode the audio signal. (io) For practical reasons, the signals are sampled at a rate well above Nyquist rate at 44100 samples per second. If L = 65,536, determine number of bits per second required to encode the signal and transmission bandwidth of encoded signal. Solution : (i) To obtain Nyquist rate ‘The bandwidth of the signal is, W = 15 kHz. 2w 2«15KHz = 30 KHz Nyquist rate a "Digital Communications 1-94 Pulse Digital Modulation (i) To determine number of bits Number of levels,q = L = 65,536 Hence binary digits required to encode each sample will be, qz2 or 2 = logeq log » 65536 = 16 bits (iil) To determine signaling rate v = 16 bits/sample are used. The samples are taken at the rate of 30,000 samples/sec. Hence signaling rate will be, reo = 16x 30,000 = 480 k bits/sec (iv) To obtain By if f, = 44.1 kHz Levels used are q = 65,536 * v = logaq = 16 bits f, = 44100 samples/sec From equation 1.85, transmission bandwidth required to encode the signal will be, 4 By = $x 16% 44100 : 352.8 kHz tm Example 1.8.14 : The output signal to noise ratio of « 10 bit PCM was found to be 30 4B. The desired SNR is 42 dB. It was decided to increase the SNR to the desired value by increasing the number of quantization levels. Find the fractional increase in transmission bandwidth required for this increase in SNR. Sol. ; (0) To obtain no of bits for 42 dB Signal to noise ratio of PCM is given as, S\ip = (a}e = (48 + 60) dB Above equation shows that signal to noise ratio increases by 6 dB with every bit. It is given that s x = 3048 for 10 bitsDigital Communications 1-98 Pulse Digital Modulation Theory Questions 1. With the help of neat diagrams, explain the transmitter and receiver of pulse code modulation 2. What is uniform (linear) quantization ? 3. Expalin quantization error and derive an expression for maximum signal to noise ratio in PCM system that uses linear quantiza 4. Derive the relations for signaling rate and transmission bandwidth in PCM syste. 5. What is the necessity of nonuniform quantization and explain companding ? Unsolved Examples 1. A 40 MB huard disk i used to store PCM date. The signal is sampled at 8 kHz and the encoded PCM is to have an average signal to noise ratio of at least 30 dB. For how many minutes the PCM deta can be stored on the hard disk ? Ans. : 133 min} 2. In the binary PCM system, find out the minimum number of bits required so that quantizing noise is less than +k percent of the analeg level. Ana. 202 10g2(50/ Rl 3. The Gaussien distrituted random variable with zero mean and unit variance 1s applied to the input of uniform quantizer. (a) What is the probability that the amplitude of this input lies outside the range 44 ? (b) Using the resuit of past (a), find out the signal to quantization noise ratio. (Ans. (a) Lin 104 (b) (S/N) AB = 60 -7.2dB) 1.9 Digital Multiplexers Digital signals are the sequences of binary 1 and 0 symbols. Digital multiplexing technique simultaneously transmits the symbols from many channels by interleaving them This is very much similar to time division multiplexing. In digital multiplexing there are no constraints like periodic sampling and waveform preservation. The digital multiplexing uses a binary multiplexers and their hierarchies. A binary multiplexer merges input bit from different sources into one signal for transmission via a digital communication system. The multiplexing of various digital signals can be bit by bit or by words or by characters.Additional pulses are inserted in the multiplexed data stream to identify the different channels or frames. These are called control bits. The multiplexer performs following operations. 1. Establish the frame as the smallest time interval containing at least one bit from every input. 2. Assign to each input a number of unique bit slots within the frame. Insert control bits for frame identification and synchronization. Make allowance for any variations of the input bit rates.Digital Communications 1-97 Pulse Digital Modulation 1.9.4 Types of Digital Multiplexers ‘There are following types of digital multiplexers Synchronous multiplexers : When single master clock governs all sources synchronous multiplexers are used. Since a single master clock is used there are no bit rate variations. Synchronous multiplexing has highest throughput efficiency. Synchronous multiplexer have the increased complexity because of master clock signal. Asynchronous multiplexers : Asynchronous multiplexers are used for digital data sources that operate in a start/stop mode producing bursts of characters with variable spacing between the bursts. Buffering and character interleaving makes it possible to merge these sources into a synchronous multiplexed bit stream. Quasi synchronous multiplexers : Quasi synchronous multiplexers are used when input bit rates have the same nominal value but vary within specified bounds. These multiplexers arranged in a hierarchy of increasing bit rates, constitute the building blocks of interconnected digital communication system. 1.9.2 Multiplexing Hierarchies Fig. 1.9.1 shows the multiplexing hierarchy for digital communication. =e ne Voieo — 15 Mbls $e Tank wom ot k's m Furst level Voice PCM 64 kbis A 274 Mbis Fig. 1.9.1 Multiplexing hierarchy for digital communication In this hierarchy the third level is used, for multiplexing purposes and other three levels are designed for point to point transmission and multiplexing. The bit rate at the next level is more than the sum of all the channels multiplexed at the input of that level. Table 1.9.1 shows inputs and rates for a typical digital multiplexer.Digital Communications 1-98 Pulse Digital Modulation Lovels Number of inputs Output bit rate or bits por sec. First levet 24 15x 108 Second leve! 4 ox1e Third level 7 44x 108 Fourth level 6 7h 108 Table 1.9.1 Inputs and rates of digital multiplexers for AT and T system From the above table we can see that fourth level of the multiplexer has total number of inputs as, Multiplexed inputs to 4! level 2x 4x7 x6 4032 Voice PCM signals Since 4" Jevel is the last level, the digital multiplexer multiplexes total 4032 voice PCM signals. The bit rate of the fourth level is 274X108 bits per second. That is, it is the final output signaling rate, 1 = 274x106 We know that transmission channel bandwidth B; should be, ip Bf? 2 137 MHz 1.9.3 PCM TDM System 1.93.1 Multiplexing Hierarchy ‘The PCM-TDM system uses many codecs as shown in Fig. 1.92. The codec is basically a PCM encoder (transmitter) and decoder (receiver). Codec generates serial stream of PCM data. At the receiver side, codec receives serial PCM data and generates analog signal. The sampling frequency of PCM can be selected by external clock given to the codec. One codec per channel is used. The outputs from various codecs are combined by the multiplexer into single bit stream. This bit stream is converted to baseband waveform by line waveform generator. The low pass filter (LPF) bandlimits the baseband signal. The waveform regenerator is used at the receiver to construct the input noisy waveform to clear digital signal. The Demux then detects individual channel signals and separates them. The codecs then recover the required analog signal.Digital Communications. 1-99 Pulse Digital Modulation ‘analog ‘put 4 Codec * Tecu} Multiplexed cata | 1 2 Codec t * 1 ° wr L__] ; N Codee i 7M : Channel Put a) PF regenerator Fig, 1.9.2 TDM/PCM system 1.93.2 Multiple Channel Frame Alignment For TDM / PCM (1) System) The multiple channel alignment is very important in TDM/PCM system. Fig. 1.9.3 shows the TDM frame format of most widely used T1 system. FI ae seb susrore [1]? [3 [2] 1617] 8] Cranes ccna cruel ret Eola Dele ee Ps 1 PEEEPELEPEEEP PAPE EEL aa roe TEEEEEPELEDEEEPELEPLELE EK PEED ___ 2 1 PPLE 13 does Fig, 1.9.3 Multiple channel frame alignment in T1 systemDigital Communications As shown in the Fig.1.93, this system contains a multiframe of 12 frames. The duration of the multiframe is 1.5 msec. Each frame consists of samples from 24 channels. Thus the samples of 24 channels are Time Division Multiplexed. Each channel sample is encoded into 8 bits. Thus the total bits of 24 channels will be 24x8=192 bits. This indicate the start of the next frame, the frame sync bit or 'S' bit is transmitted at the begining of each frame. Thus the total bits in one frame are (24x8) +1=19 bits. Calculation of bit rate : Each channel is normally sampled at 8 kHz rate. Thus the time between any two successive samples of single channel will be gag 7-=125 microseconds, In the TDM system, the samples from each channel is transmitted in each successive frame. Hence the duration of the frame will also be 125 microseconds. This is shown in Fig, 1.9.3. Bits per frame = 24 channels/frame x 8 bits/channel + 1 frame sync bit = 193 bits Number of bits per frame 193 bits Bit rate R, = Numberof bitsperframe | __193 bits _ rate Ky “Time of one frame 125% 10° seconds = 1,.544%10° bits/sec ‘The signaling information is transmitted by replacing the 8" bit (ie. LSB) in each channel by signaling bit in every sixth frame. Thus, Signaling period = Period of the signaling bit = 125x10% x6 = 750x10° sec 1 ‘signaling period 1 750x106 = 1.333 Kbps. Signaling rate = ‘> Example 1.9.1: The T; carrier system used in digital telephony multiplexes 24 voice channels based on 8 bit PCM. Each voice signal is usually put through a lowpass filter with cutof frequency 3.4 kHz The filtered signal is sampled at 8 kHz. In addition a single bit is added at the end of frame for the purpose of synchronization.Digital Communications 4-104 Pulse Digital Modulation Calculate i) The duration of each bit ii) The resultant transmission rate iii) Minimum required transmission bandwidth. Solution : i) Duration of each bit ‘The T; system is explained just now. The signals are sampled at 8 KHz. Hence the time between any two successive samples of the same channel will be <1. = 125 ps. ‘5000 Fig. 1.9.3 shows the structure of the frame. In one frame of 125 pis, the total bits are, Bits per frame = 24 channels/frame x 8 bits/channel + 1 frame sync bit 193 bits ‘Time duration of one frame Time duration of one bit = Bits per frame _ 12x 10-8 7 793 0.6476 sec/bit ii) Transmission rate ‘The transmission rate is the bit rate K,, which is the reciprocal of duration of one bit ie., ——_!_ Duration of one bit 1 © 0.6476x 10% Ry = = 1544x108 bits/sec iii) Transmission bandwidth ‘The transmission bandwidth in PCM must be greater than half of the bit rate. i.e, 1 Br BFR, > birstoent Br 2 772kHzDigital Communications 4-402 Pulse Digital Modutation vm Example 1.9.2: Twenty four voice channels of 4 kHz bandwidth each sampled at Nyquist rete and encoded into 8 bit PCM are time division multiplexed with 1 bitframe as synchronization bit. What is bit rate at the output of multiplexers? Solution : Given data is N = 24 channels Bandwidth Wo = 4 kHz o = 8 bits Nyquist rate = 2W = 204 kHz = BkHz ‘Twenty four voice channels are transmitted in one frame-Each of the channel has 8 bits. One bit is added in every frame for synchronization. Hence, Bits in one frame = 24x8-+1 ( Synchronization bit ) 193 bits / frame. Frames are transmitted at the nyquist rate, Hence bit rate at the ouput of multiplexer will be, Bit rate = number of bits/frame x number of frames/sec. 193 x 8000 1.544 x 10° bits/sec. ‘Thus the bit rate is 1,544 mbps. mma Example 1.9.3 : Describe the digital multiplexing of number of telephone channels, data channels, TV channels. Draw an appropriate diagram showing different multiplexing levels of either AT and T or CIT standard. Solution : Fig. 1.94 shows the configuration of digital multiplexer of AT & T standard. In the first level PCM voice channels and digital data channels are multiplexed. The second level multiplexes T; signal and visual telephone data. The maximum bit rate of the forth multiplexing level can be 274 x 10° bits per second.Digital Communications 1-103 ee, Ty_1.5 Mois ico ‘»|Channot telephone f™"| Bank chennels —>} Pulse Digital Modulation mz bo 274 Mbls 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. Explaon 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. (¥) Encryption or decryption can be easily incorporated for security purpose. Limitations of PCM (i) PCM systems are complex compared to analog pulse modulation methods. (ji) The channel bandwidth is also increased because of digital coding of analog pulses.04 Digital Communication: Digital Modulation Modifications of PCM () PCM can be modified to delta modulation. It is more simplified method of implementation. (i) The PCM can be used in wideband communications channels to overcome the bandwidth problem. (ii) With the help of data comparison along with PCM, the redundancy can be removed and data rate can be reduced. 1.11 Differential Pulse Code Modulation 1.11.1 Redundant Information in PCM ‘The samples of a signal are highly corrected with each other. This is because any signal does not change fast. That is its value from present sample to next sample docs not differ by large amount. The adjacent samples of the signal carry the same information with little difference. When these samples are encoded by standard PCM system, the resulting encoded signal contains redundant information. Fig. 1.11.1 illustrates this. Fig. 111.1 shows a continuous time signal x() by dotted line. This signal is sampled by flat top sampling at intervals T,, 27; , 37; ...T,. The sampling frequency is selected to be higher than nyquist rate. The samples are encoded by using 3 bit (7 levels) PCM. The sample is quantized to the nearest digital level as shown by small xt) bits (evets) 7411) 110) 6 (110). 101) (101) 5 (101). g 4(100} 30011) 2(010) 4001) 0 xor) Fig. 1.11.1 Redundant information in PCMDigital Communications 1-105 Pulse Digital Modulation circles in the diagram. The encoded binary value of each sample is written on the top of the samples. We can see from Fig. 1.11.1 that the samples taken at 47, ,5T, and 67, are encoded to same value of (110). This information can be carried only by one sample. But three samples are carrying the same information means it is redundant. Consider another example of samples taken at 97, and107,. The difference between these samples is only due to last bit and first two bits are redundant, since they do not change. 1.11.2 Principle of DPCM If this redundancy is reduced, then overall bit rate will decrease and number of bits required to transmit one sample will also be reduced. This type of digital pulse modulation scheme is called Differential Pulse Code Modulation. 1.11.3 DPCM Transmitter ‘The differential pulse code modulation works on the principle of prediction. ‘The value of the present sample is predicted from the past samples. The prediction may not be exact but it is very close to the actual sample value. Fig. 1.11.2 shows the transmitter of Differential Pulse Code Modulation (DPCM) system. The sampled signal is denoted by x(nT,) and the predicted signal is denoted by £(nT,). The comparator finds out the difference between the actual sample value x(n T,) and predicted sample value 3(nT,). This is called error and it is denoted by ¢(nT,} It can be defined as, e(nT,) = x(nT,)-3(nT,) (LI) Comparator Sampea input x(0T,) DPCM signal TY Fig. 1.11.2 Differential pulse code modulation transmitter Thus error is the difference between unquantized input sample x(wT,) and prediction of it 3(nT,). The predicted value is produced by using a prediction filter. The quantizer output signal e,(n7,) and previous prediction is added and given asDigital Communications 4-106 Pulse Digital Modulation input to the prediction filler, This signal is called a, (,). This makes the prediction more and more close to the actual sampled signal. We can see that the quantized error signal ¢, (nT,) is very small and can be encoded by using small number of bits. Thus number of bits per sample are reduced in DPCM. ‘The quantizer output can be written as, eg (Ts) = eT) +q(nT,) ve (111.2) Here q(T,) is the quantization error. As shown in Fig. 1.112, the prediction filter input x, ("7,) is obtained by sum i(nT,) and quantizer output i. xy(nT,) = RnT,)eq (nT) w- (11.3) Putting the value of e, (nT,) from equation 1.11.2 in the above equation we get, X_(nT.) = FnT)+e(nT,)+q(nT.) (L114) Equation 1.11.1 is written as, e(nT,) = x(nT,)~*(nT,) e(eT)+3(nT,) = x(nT,) (L115) 2. Putting the value of e (1 T,) + #(1T,) from above equation into equation 1.114 we get, x,(nT) = x(nT.)+q(0T,) w= (L116) Thus the quantized version of the signal x, (1 J.) is the sum of original sample value and quantization error q(nT,). The quantization error can be positive or negative. Thus equation 1.11.6 does not depend on the prediction filter characteristics. 1.11.4 Reconstruction of DPCM Signal Fig, 1.11.3 shows the block diagram of DPCM receiver. pcm, ren Decoder > Outeut Fig, 1.11.3 DPCM receiver ‘The decoder first reconstructs the quantized error signal from incoming binary signal. The prediction filter output and quantized error signals are summed up to give the quantized version of the original signal. Thus the signal at the receiver differs from actual signal by quantization error q(nT;), which is introduced permanently in the reconstructed signal. gag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(() is approximated to step signal by the delta modulator. This step size is fixed. The difference between the input signal x(t) and staircase approximated signal confined to two levels, ic +8and-8 If the difference is positive, then approximated signal is increased by one step ie. 6. If the difference is negative, then approximated signal is reduced by “é. When the step is reduced, ‘0’ is transmitted and if the step is increased, ‘I’ 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. rt 1 f | to) + tht i nk Le Tye i @ | {. ‘Sampling | Perel “sores | CAT | pet | Binary one | Jody sltsecen ifertie ojo} el modulation waveform (2-1)Digital Communications 2-2 Delta Modulation The principle of delta modulation can be explained by the following set of equations. The error between the sampled value of x(t) and last approximated sample is given as, e(nT,) = x(rT,)-3(nT,) s+ (211) Here, e(nT,) = Error at present sample x(nT,) = Sampled signal of x(t) H(nT,) = Last sample approximation of the staircase waveform. We can call u(w7) as the present sample approximation of staircase output. Then, u[(r-1)T,] = F(T) wee (2.1.2) = Last sample approximation of staircase waveform. Let the quantity b (nT,) be defined as, b(nT,) = 5 sgnle(nT,)] s+ (213) That is depending on the sign of error e(mT,) the sign of step size 5 will be decided. In other words, bw) = +8 if xT) 2 RT) =-3 if -x(nT) < XMnT,) wes (214) Mo b(nT,) = +8; binary ‘1’ is transmitted and if b(nT,) 7, = Sampling interval. 8; binary ‘0’ is transmitted. 2.1.2 DM Transmitter Fig. 2.1.2 (@) shows the transmitter based on equations 2.1.3 to 2.1.5. The summer in the accumulator adds quantizer output (£5) with the previous sample approximation. This gives present sample approximation. ic, u(nT,) =u(nT, -T,) +148] oF = ul(a-T,]+b(nT,) e+ Q15) ‘The previous sample approximation u{(n-1)T,] is restored by delaying one sample period T,. The sampled input signal x(nT,) and staircase approximated signal 2(nT,) are subtracted to get error signal e(nT,).Digital Communications 2-3 Delta Modulation Sampled g(t) el input te TS xT) ‘Aecamaiaior © Accumulator te) Fig. 2.1.2 (a) Delta modulation transmitter and (b) Delta modulation receiver Depending on the sign of e(nT,) one bit quantizer produces an output step of +8 or ~8. If the step size is +8, then binary ‘I’ is transmitted and if it is ~5, then binary ‘0’ is transmitted. 2.1.3 DM Recaiver At the receiver shown in Fig. 2.12 (b), the accumulator and low-pass filter are used. The accumulator generates the staircase approximated signal output and is delayed by one sampling period T,. It is then added to the input signal. If input is binary ‘1’ then it adds +6 step to the previous output (which is delayed). If input is binary ‘0’ then one step '8' is subtracted from the delayed signal. The low-pass filter has the cutoff frequency equal to highest frequency in (0). This filter smoothen the staircase signal to reconstruct x(t).Digital Communications 2-4 Delta Modulation 2.2 Advantages and Disadvantages of Delta Modulation 2.2.1 Advantages of Delta Modulation ‘The delta modulation has following advantages over PCM, 1. Delta modulation transmits only one bit for one sample. Thus the signaling rate and transmission channel bandwidth is quite small for delta modulation. 2. The transmitter and receiver implementation is very much simple for delta modulation. There is no analog to digital converter involved in delta modulation. 2.2.2 Disadvantagos of Dolta Modulation Granular noise Slope - overioad distortion Staircase approximation: ut) J f Fig. 22.1 Quantization errors in delta modulation ‘The delta modulation has two drawbacks - 2.224 Slope Overload Distortion (Startup Error) This distortion arises because of the large dynamic range of the input signal. ‘As can be seen from Fig. 22.1 the rate of rise of input signal x(f) is so high that the staircase signal cannot approximate it, the step size ‘6’ becomes too small for staircase signal u(t) to follow the steep segment of x(\). Thus there is a large error between the staircase approximated signal and the original input signal x(i). This error is called slope overload distortion. To reduce this error, the step size should be increased when slope of signal of x(0) is high. Since the step size of delta modulator remains fixed, its maximum or minimum. slopes occur along straight lines. Therefore this modulator is also called Linear Delta Modulator (LDM). 2.22.2 Granular Noise (Hunting) Granular noise occurs when the step size is too large compared to small variations in the input signal. That is for very small variations in the input signal, the staircaseDigital Communications 2-5 Detta Modulation signal is changed by large amount (5) because of large step size. Fig. 2.2.1 shows that when the input signal is almost flat, the staircase signal u()) keeps on oscillating by +5 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. my 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 tinear 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 215 as, unt.) = uf(n-1)T, +n) w= (221) Here BnT,) = £8 or Ssgn[aXnT,)] Thus WnT,) basically represents error signal. Sign of step size ‘8 depends upon whether e(1T,) is positive or negative. Now we will show that the signal can be reconstructed only with the help of encoded error signal, i.e. K(nT,). The accumulator of Fig. 21.2(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, uin) = x(n 1) +n) (22.2) Observe that this is recursive equation. Hence u(7t~ 1) can be calculated as, un =1) = x(n 2) +H ~1) (223) Hence equation 2.2.2 becomes, un) = un ~ 2) +(n—1) +O(n) w= (22.4) From equation 22.3 we can calculate 1(n1—2) as, u(n=2) = (2-3) +b(n-2) Hence equation 2.2.4 becomes, un) = u(t 3)+b(n - 2) +0(n -1) + K(n)Digital Communications 2-6 Delta Modulation Above equation can be generalized as, n(n) = Wn) +001 -1) +n 2)... Thus 1(n) can be reconstructed totally from encoded errors b(n), (1-1), 2.3 Adaptive Delta Modulation 2.3.1 Operating Principle ‘To overcome the quantization errors due to slope overload and granular noise, the step size (6) is made adaptive to variations in the input signal x(¢). Particularly in the steep segment of the signal x(t), the step size is increased. When the input is varying slowly, the step size is reduced. Then the method is called Adaptive Delta Modulation (ADM). ‘The adaptive delta modulators can take continuous changes in step size or discrete changes in step size. 2.3.2 Transmitter and Receiver Fig, 23.1 (a) shows the transmitter and 23.1 (b) shows receiver of adaptive delta modulator. The logic for step size control is added in the diagram. The step size increases or decreases according to certain rule depending on one bit quantizer output. Logie for step size }#— ‘control +} output 5 —> variable Fig, 2.3.1 Adaptive delta modulator (a) Transmitter (b) ReceiverDigital Communications 2-7 Delta Modulation For example if one bit quantizer output is high (1), then step size may be doubled for next sample. If one bit quantizer output is low, then step size may be reduced by one step. Fig. 2.3.2 shows the waveforms of adaptive delta modulator and sequence of bits transmitted. In the receiver of adaptive delta modulator shown in Fig. 2.3.1 (b) the first part generates the step size from each incoming bit. Exactly the same process is followed as that in transmitter. The previous input and present input decides the step size. It is then given to an accumulator which builds up staircase waveform. The low-pass filter then smoothens out the staircase waveform to reconstruct the smooth signal. fF enmmertgasrs| +] +] 1} 1 fo of of of Fig. 2.3.2 Waveforms of adaptive delta modulation 2.3.3 Advantages of Adaptive Delta Modulation Adaptive delta modulation has certain advantages over delta modulation. i 1. The signal to noise ratio is better than ordinary delta modulation because of the reduction in slope overload distortion and granular noise. 2. Because of the variable step size, the dynamic range of ADM is wide. 3. Utilization of bandwidth is better than delta modulation. Plus other advantages of delta modulation are, only one bit per sample is required and simplicity of implementation of transmitter and receiver. im Example 2.3.1 : Consider a sine wave of frequency f,, and amplitude Ay, applied to a delta modulator of step size 8. Show that the slope overload distortion will occur if e,Digital Communications 2-8 Delta Modulation > Bp Te Am > ‘where T, is the sampling period. (Nov/Dec-2004, 4 Marks) Solution : Let the sine wave be represented as, x) = Ay sinQnfy Slope of x(t) will be maximum when derivative of x(t) with respect to ‘t’ will be maximum. The maximum slope of delta modulator is given from Fig. 2.1.1 as, Step size Max. slope = =——*——__ 'P® = Sampling period o- (Q3.1) Slope overload distortion will take place if slope of sine wave is greater than slope of delta modulator i.e. sal) > ta g mnan| Ay sin fy 5 > lo maK|Ay, 28 fy C28 2% fy 1) > Ble Blo Slo An 2fin > wu (23.2) im Example 2.3.2 : A delta modulator system is designed to operate at five times the Nyquist rate for a signal with 3 kHz bandwidth. Determine the maximum amplitude of 2 kHz input sinusoid for which the delta modulator does not have slope over load. Quantizing step size is 250 mV. Derive the formula that you use. Solution : In example 23.1 we have derived the relation for slope overload distortion which will occur if, 8 An > aT By equation 23.2 ow (2.3.3)Digital Communications 2-10 Delta Modulation fe} Fig. 2.3.3 Delta modulator Solution : Optimum step size means slope overload will not occur. Please refer to the relationship proved in example 2.3.1 It is given as, slope overload distortion will occur if, An > GEE Here 4m = A is the amplitude of sine wave & = kis the step size T= Ais the sampling duration. Hence above equation can also be written as, slope overload distortion will occur if, k A> RTE w= (23.5) Here the sine wave is given as, mt) = O.1sin(2n x10°/) = Asin (2n x fn!) Hence A=01V Jim = 109 Hz and fe = 2108 Hz For k= 4 mv From equation 235, slope overload will occur if k A> TahyThDigital Communications 2-1 Delta Modulation 4x10% > 1x10" __ 2nx103 / (2x104) > 0.01273 We know that A is 0.1 V which is greater than 0.01273. Hence slope overload distortion will occur for step size of 4 mV. For k = 60 mV From equation 2.35 slope overload will occur if K A? WTR 60x10 2nx103 / (2%104) > 0.19098 Since A is 0.1 V, the slope overload distortion will not ocur in this case. This is because A(0.1 V) is less than 0.19098 Y. Hence slope overload distortion will not occur for step size of 60 mV. Sr. No. Step size k Slope overload distortion 1 mv ‘Slope overtoad distortion occurs 2. 60 mv Slope overload distortion will not occur Table 2.3.1 Results wm Example 2.3.4 : Find the signal amplitude for minimum quantization error in a delta modulation system if step size is 1 volts having repetition period 1 ms. The information signal operates at 100 Hz. Solution: The quantization error is minimum when slope overload distortion is absent. Then the quantization error is due to granular noise only. We know that the slope overload distortion is absent if, 8 jn SEE Here Ay, is the signal amplitude 8 is step size. Given 1V fn is signal frequency. Given 100 Hz T, is sampling duration. Given 1 ms.Digital Communications 2 Delta Modulation Hence signal amplitude becomes, ——1 2nx 1001x109 < 16V A ie. Am That is, the signal amplitude should be less than 1.6 volts to have minimum quantization error. wa Example 2.3.5 : Derive an expression for signal to quantization noise power ratio for delta modulation. Assume that no slope overload distortion exists. [March-2006, 16 Marks] Solution : (i) To obtain signal power : In example 23.1 we have derived that slope overload distortion will not occur if a mar, Here A,, is peak amplitude of sinusoided signal 5 is the step size fry is the signal frequency and T, is the sampling period. From above equation, the maximum signal amplitude will be, s .3 An = it ww (236) Signal power is given as, =v Poe Here V ia the rms value of the signal. Here V = “8. Hence above equation becomes, ‘A, p= (Se) 7R (& Normalized signal power is obtained by taking R = 1. Hence, p= 42 2Digital Communications 2-13 Delta Modulation Putting for A,,from equation 23.6, 8 P= mee wn (23.7) This is an expression for signal power in delta modulation. Gi) To obtain noise power 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 [-8,5} This is shown in Fig. 23.4 From this figure the PDF of quantization error can be expressed as, Fig. 2.3.4 Uniform distribution of quantization error 100 0 for e Example 23.9: Ina single integration DM scheme the voice signal is sampled at a rate of 64 Kidz. The maximum signal amplitude is 2 volts. Voice signal bandwidth is 35 kHz. Determine the minimum value of step size to avoid slope over load and calculate granular noise power. Solution ; The given data is, h 1 * 64x10 OxlHz +. T; > 1 av W =35 kHz = 3500 Hz "Digital Communications 2-19 Delta Modulation i) To determine step size (5) Slope overload distortion will not occur if, 3 = S OR Putting values, 2 < 5 t 2nx3500x—t eaxi03 ‘ 5 2 0.687 volts. li) To calculate noise power Noise power is given by equation 2.3.11 as, 7 1 wrs2 _ G03 3 3 8.6 mW. (0687)? Noise power im Example 2.3.10: In a single integration DM scheme, the voice signal is sampled at a rate of 64 kHz. The maximum signal amplitude is 1 volt, voice signal bandwidth is 3.5 kHz. i) Determine the minimum value of step size to avoid slope overload. i) Determine granular noise N, iii) Assuming signal to be sinusoidal, calculate signal power and signal to noise ratio. iv) Assuming that noise signal amplitude is uniformly distributed in the range (AA, 1) determine the signal power and signal to noise ratio. (Nov2005, 16 Marks} Solution : The given data is, fe 4kHZ, Ay =1V, W=35 kHz. 1) To obtain step size Slope overload will not occur if, Putting values in above equation with 1 T= Gauge and fy = 35 KixDigital Communications Delta Modulation é 2mx 350% 1s axe 5 > 0.436 volts ii) To obtain granular noise powor Noise power is given by equation 2.3.11 as, WT,5? _ W8? Ne 3. = 3.5% 103 x(0.3436)? 3x 64x105 = 215 mW Ill) To obtain SIN ratio Amplitude of the signal is 1V. For sinusoidal signal, the signal power will be 2 Hence signal to noise ratio will be, S _ Signalpower_ 1/2 P mal power __1/2 _ _ 999: N 7 Noisepower ~ aisxio3 7 223 = 10log yo & = 23.66 4B g zo a f iv) To calculate ratio if signal is uniformly distributed over the range (-1, 1) 10 Fig. 2.36 shows the pdf of the 4 signal. It can be easily calculated that fx) } if itis distributed over (-1, 1). * Hence mean square value of the signal can be calculated as, Fig. 2.3.6 Uniformly distributed signalDigital Communications Delta Modulation 7 Normalized signal power = x =X? withR=1 =i =4W Hence signal to noise ratio becomes, 1 ‘S _ Signalpower_ 3 N * Noise power ~ 235x10> or (3) = 1log 39155 = 21.9 dB te Theory Questions 1. Explain delta modulation in detail suitable diagram. Explain ADM and conpare 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 maxinsun power that nay be transmitted without slope overload distortion ? tans. Fy Safle 2.4 Comparison of Digital Pulse Modulation Methods ‘Table 2.4.1 shows the comparison of FCM, 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.Passband Data Transmission 3.1 Introduction ‘There are basically two types of transmission of digital signals : 1) Baseband data transmission : The digital dala is transmitted over the channel directly. There is no carrier or any modulation. This is suitable for transmission over short distances. 2) Passband data transmission : The digital data modulates high frequency sinusoidal carrier. Hence it is also called digital CW modulation. It is suitable for transmission over long distances. 3.1.1 Types of Passband Modulation The digital data can modulate phase, frequency or amplitude of carrier. This gives rise to three basic techniques : 1) Phase shift keying (PSK) : In this technique, the digital data modulates phase of the carrier. 2) Frequency shift keying (SK) : In this technique, the digital data modulates frequency of the cartier. 3) Amplitude shift keying (ASK) : In this technique, the digital data modulates amplitude of the cartier. 3.1.2 Types of Reception for Passband Transmission ‘There are two types of methods for detection of passband signals. 1. Coherent (Synchronous) detection : In this method, the local cartier generated at the receiver is phase locked with the carrier at the transmitter. Hence it is also called synchronous detection. 2. Noncoherent (Envelope) detection : In this method, the receiver carrier need not be phase locked with transmitter carrier. Hence it is also called envelope detection, Noncoherent detection is simple but it has higher probability of error. @-1)Digitat Communication 3-2 Passband Data Transmission 3.1.3 Requirements of Passband Transmission Scheme Any passband transmission scheme should satisfy following requirements = 1. Maximum data transmission rate 2. Minimum probability of symbol error. 3. Minimum transmitted power. 4, Minimum channel bandwidth. 5. Maximum resistance to interfering signals. 6. Minimum circuit complexity. 3.1.4 Advantages of Passband Transmission over Baseband Transmission 1, Long distance transmission. 2. Analog channels, can be used for transmission. . Multiplexing techniques can be used for bandwidth conservation. 4. Problems such as ISI and crosstalk are absent. Passband transmission can take place over wireless channels also. 6. Large number of modulation techniques are available. Drawbacks of Passband Modulation 1. Modulation and demodulation equipments, transmitting/receiving antennas, interference problems make the system complex. 2. It is not suitable for short distance communication. 3.1.5 Passband Transmission Model Fig. 3.1.1 shows the model of passband data transmission system Trensmiter Receiver Fig. 3.1.1 Model of passband data transmission systemDigital Communication 3-3 Passband Data Transmission ‘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 s(t) is energy signal 4, Channel : The modulated signal s(t) is transmitted over the communication channel + The channel is assumed to be linear and of enough bandwidth to accommodate the signal s(t). N + The channel noise is white Gaussian of zero mean and psd of —? 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 jon of the transmitted signal. The effect of channel noise is minimized and correct estimate of symbol jr is obtained. 3.2 Binary Phase Shift Keying (BPSK) ‘* In binary phase shift keying (BPSK), binary symbol ‘I’ and ‘0 modulate the phase of the carrier. Let the carrier be, s() = Acos(2r fo!) + (3.2.1) ‘A’ represents peak value of sinusoidal carrier. In the standard 10 load register, the power dissipated will be, pe daa A = JP es (3.2.2) ‘* When the symbol is changed, then the phase of the carrier is changed by 180 degrees (x radians). * Consider for example, Symbol ‘I’ = 51 () = ¥2P cos (2x fy t) (32.3)Digital Communication 3-4 Passband Data Transmission if next symbol is '0' then, Symbol ‘0 = s» (!) = V2P cos (2nfyt +m) oo 3.24) Since cos (9 +m) = —cos®, we can write above equation as, $2 () = ~V2P cos (2n fo) w+ (3.2.5) With the above equation we can define BPSK signal combinely as, 8(t) = b(®) V2P cos (2x fy t) (3.2.6) Here b(t) = +1 when binary '1' is to be transmitted “1 when binary '0' is to be transmitted 3.22 Graphical Representation of BPSK Signal Fig. 3.2.1 shows binary signal and its equivalent signal 6 (0 signal Fig. 3.2.1 (a) Binary sequence (b) Its equivalent bipolar signal b(¢) (c) BPSK signal ‘As can be seen from Fig, 3.2.1 (b), the signal b(t) is NRZ bipolar signal. This signal directly modulates cartier cos (2nfy 9.Digital Communication 35 Passband Data Transmission 3.2.3 Generation and Reception of BPSK Signal Nov./8e~ 2005 3.23.1 Generator of BPSK Signal BPSK ‘signal Carer signal Fig. 3.2.2 BPSK generation scheme + The BPSK signal can be generated by applying carrier signal to the balanced modulator + The baseband signal b(#) is applied as a modulating signal to the balanced modulator. Fig. 3.2.2 shows the block diagram of BPSK signal generator. + The NRZ level encoder converts the binary data sequence into bipolar NRZ signal. 3.23.2 Reception of BPSK Signal Fig, 3.2.3 shows the block diagram of the scheme to recover baseband signal from BPSK signal. The transmitted BPSK signal is, 3() = b(t) V2P cos Qn fy t) BPSK signal cos*(2rfo! +0) cosaiantet +0) from channel Frequency] ‘iwder by two casiQalst + 0) 4 eso wy VB eosteno) | P | wi FB cos2ainn s [ann] synchronizer Fig. 3.2.3 Reception BPSK scheme ‘Synchionous| cerrodulator (rnuttiplien)Digital Communication 3-6 Passband Data Transmi Operation of the receiver 1) Phase shift in received signal : This signal undergoes the phase change depending upon the time delay from transmitter to receiver. This phase change is normally fixed phase shift in the transmitted signal. Let the phase shift be 8. Therefore the signal at the input of the receiver s() = b (0 V2P cos (2m fot +0) we (3.2.7) 2) Square law device : Now from this received signal, a carrier is separated since this is coherent detection. As shown in the figure, the received signal is passed through a square law device. At the output of the square law device the signal will be, cos? (2ref t +8) Note here that we have neglected the amplitude, because we are only interested in the carrier of the signal We know that, costg = 1te0s 20 L4-c0s 2(2x fy t +0) ST Oxf t-+9) or 2(2e fy t +0) Hore } represents a DC level. 3) Bandpass filter : This signal is then passed through a bandpass filter whose passband Is centered around 2g. Bandpass filter removes the DC level of and at its output we get, 082 2m fot +0) This signal has frequency of 2f, 4) Frequency divider : The above signal is passed through a frequency divider by two. Therefore at the output of frequency divider we get a carrier signal whose frequency is fy i.e. cos (2n fo t +0). 5) Synchronous demodulator : The synchronous (coherent) demodulator multiplies the input signal and the recovered carrier. Therefore at the output of multiplier we get, b(t) JP cos(2x fy t+) x cos (2 fyt +0) = b(t) V2P cos? (2m fo t +0) " BOVE x FU +008 2028 fy +0)Digital Communication 3-7 Passband Data Transmission =b 5 [+ cos 2(2n fy t +0)1 w= (3.2.8) 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 $2 temperorily. This connects the output of an integrator to the deci It is equivalent to sampling the output of integrator. on device, * The synchronizer then opens switch S, and switch S, is closed temperorily. This resets the integrator voltage to zero. The integrator then integrates next bit Let us assume that one bit period 'T,' contains integral number of cycles of the cartier. That is the phase change oc crossing. This is shown in of sinusoidal carrier. rs in the carrier only at zero 3.2.1 (c). Thus BPSK waveform has full cycles To show that output of integrator depends upon transmitted bit + In the All bit interval we can write output signal as, p kt So (kT,) = san fe J Wr cos22n fy t +00] at «bn from equation 32.8 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 equation as, Bp oa 1 So (KT) = sane fia J cos 22x fy t r0) at | “ J oT aT At Here [cos (2x fy t +8) dt =0, because average value of sinusoidal waveform is wen zero if integration is performed over full cycles. Therefore we can write above equation as, pt sy kT) = san fe foiat (DTDigital Communication 3-8 Passband Data Transmission P yk Ty ha Maya, = bk rub Te ~(k-1)T} = SAT) = VOTH) YS Ts - 629) Ud) This equation shows that the output of the receiver depends on input ie. So (RT,) a B(RT,) = b&T, Depending upon the value of b(kT;), the output so (KT) is generated in the This signal is then given to a decision device (not shown in Fig. 3.23), which decides whether transmitted symbol was zero or one. 3.2.4 Spectrum of BPSK Signals Step 1 : Fourier transform of basic NRZ pulse. We know that the waveform b(t) is NRZ bipolar waveform. In this waveform there are rectangular pulses of amplitude +V;. IF we say that each .. pulse is af around its center as shown in Tig. 324. then it becomes easy to find fourier transform of such ; pulse, The fourier transform of this Fig. 32.4 NRZ pulse type of pulse is given as, sin(n fT) 5 X() = vy, By standard relations » G20 o ° Gt) PY ) Step 2 : PSD of NRZ puise. For large number of such positive and negative pulses the power spectral density S(fis given as - EGP sa = 8211) Here X(/) denotes average value of X(f) due to all the pulses in b(t). And T, is symbol duration. Putting value of X (f) from equation 3.2.10 in equation 3.2.11 we get,Digitat Communication 3-9 Passband Data Transmission sip = nITy Step 3 : PSD of baseband signal b(t) For BPSK since only one bit is transmitted at a time, symbol and bit durations ate same ie. Ty =T,. Then above equation becomes, v2 72 (eeeuy o» (3.212) si) = VET ae The above equation gives the power spectral density of baseband signal b (). Step 4: PSD of BPSK signal. ‘The BPSK signal is generated by modulating a carrier by the baseband signal (0). Because of modulation of the carrier of frequency fp, the ‘spectral components are translated from f to fy +f and fo ~f. The magnitude of those components is divided by half. Therefore from equation 3.2.12 we can write the power spectral density of BPSK signal as, < 1fsinelfo-AT.} , 1fsinalfo +AT. | 5 = ver, eee ee a Lfsinafo + DT. aes = Vp Te {i *Gy-DT, | *2| 8G +DT The above equation is composed of two half magnitude spectral components of same frequency 'f above and below fo. Let us say that the value of £V, =+¥/P. That is the NRZ signal is having amplitudes of +/P and—JP. Then above equation becomes, sinnlf-fyTe2 1 foinw 2 Sarsk (= Te A] +} Se | 213) The above equation gives power spectral density of BPSK signal for modulating signal b(t) having amplitudes of + VP. We know that modulated signal is given by equation 32.3 and equation 32.5 as, s(t) = +V2P cos (2m fy t) since A» 2P Ifb()=+4P, then the carrier becomes OU) = VZcos(2nfyDigital Communication 3-10 Passband Data Transmission Plot of PSD + Equation 3.212 gives power spectral density of the NRZ waveform. For one rectangular pulse, the shape of $(f) will be a sinc pulse as given by equation 3.2.12. Fig. 3.2.5 shows the plot of magnitude of S(f). Fig, 3.25 Plot of power spectral density of NRZ baseband signal Above figure shows that the main lobe ranges from ~ f,, to +f,- Here f; 4 3 Since we have taken +V, =P in equation 32.12, the peak value of the main lobe is PT}. ‘* Now let us consider the power spectral density of BPSK signal given by equation 3.2.13. Fig. 32.6 shows the plot of this equation. The figure thus clearly shows that there are two lobes ; one at fy and other at — fo. The same spectrum of Fig. 3.25 is placed at +f and- fy. But the amplitudes of main a lobes are > in Fig. 3.26. Fig, 3.2.6 Plot of power spectral density of BPSK signal Thus they are reduced to half. The spectrums of S(f) as well as Sppsx (f) extends over all the frequencies.Digital Communication ae Passband Data Transmission 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) a5 well as Sypsx (f) of every channel extends over all the frequency range. + Therefore a BPSK receiver tuned to a particular cartier 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, i.e. b(t. 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 'l’ 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, St) = bt) VIP cos(2n fy) w= 215) (ii) Let's rearrange the ebove equation as, s) = 60 /PT, Fm esn 5 +» (32.16) A Git) Let 4) & 0s (2 fo f) represents an orthonormal carrier signal. Equation é 32.14 also gives equation for carrier. It is slightly different than 4, (t) defined here. Then we can write equation 32.16 as, s® = bO fPT 40 (32.17)Digital Communication 3-12 Passband Data Transmission (iv) The bit energy F;, is defined in terms of power 'P’ and bit duration 7), as, E, = PT) 32.18) Equation 3.2.17 becomes, st) = +JE, 0 (3.2.19) Here b() is simply +1.° (w/Thus on the single axis of $;() there will be two points. One point will be located at +E, and other point will be located at - JE,. is shown in Fig. 3.2.7. Represents Represents symbol 0° symbol't ~ a ~e, 4 a0 bee Fig. 3.2.7 Geometrical representation of BPSK signal At the receiver the point at +E, on 4, (#) represents symbol 'I' and point at - JE) represents symbol ‘0’. The separation between these two points represent the isolation in symbols ‘I’ and ‘0’ in BPSK signal. This separation is normally called distance ‘a’ From Fig.327 it is clear that the distance between the two points is, d = +J& -(- JE) d = 2JE, (22.20) As this distance ‘d’ increases, the isolation between the symbols in BPSK signal is more. Therefore probability of error reduces. 3.2.6 Bandwidth of BPSK Signal The spectrum of the BPSK signal is centered around the carrier frequency fy. IF fy = qh: then for BPSK the maximum frequency in the baseband signal will be fe Jy see Fig, 32.6. In this figure the main lobe is centered around carrier frequency fa and extends from fy ~ fi, t0 fo + fp- Therefore Bandwidth of BPSK signal is, BW = Highest frequency — Lowest frequency in the main lobe fo + fe ~(fo-fo) BW = 2, «= (3221)Digital Communication 3-13 Passband Data Transmission Thus the minimum bandwidth of BPSK signal is equal to twice of the highest frequency contained in baseband signal. 3.2.7 Drawbacks of BPSK : Ambiguity in Output Signal Fig. 32.3 shows the block diagram of BPSK receiver. To regenerate the carrier in the receiver, we start by squaring b(t) V2P cos(2n fy t+0). If the received signal is ~b(t) VIP cos (27 fo 1+8) then the squared signal remains same as before. Therefore the recovered carrier is unchanged even if the input signal has changed its sign. Therefore it is not possible to determine whether the received signal is equal to b(#) or -b(). This result in ambiguity in the output signal. This problem can be removed if we use differential phase shift keying. But Differential Phase Shift Keying (DPSK) also has some other problems. DPSK is given n detail in the next section, Other problems of BPSK are ISI and Interchannel interference. These problems are reduced to some extent by use of filters. mm Example 3.2.1: Determine the minimum bandwidth for a BPSK modulator with a carrier frequency of $0 MHz and an input bit rate of 500 ktps. Solkttion : The input bit rate indicates highest frequency of the baseband signal. Hence, fu = 300 Kbps 500 KHz. From equation 9.2.21, the bandwidth of the BPSK system is given as, BW = 2%, 2 x 500 kHz, 1 MHz Review Questions 1. Explain BPSK system with the help of transmitter and receiver, and state its advantagesidisadvantages over other system. 2. Derive an expression for spectrum of BPSK system and hence calculate the bandwidth required.Digital Communication 3-44 Passband Data Transmission 3.3 Differential Phaso Shift Keying (DPSK) Principle : Differential phase shift keying, (DPSK). is. differentially coherent modulation method. DPSK does not need a synchronous (coherent) carrier at the demodulator. The input sequence of binary bits is modified such that the next bit depends upon the previous bit. Therefore in the receiver the previous received bits are used to detect the present bit. 3.3.1 DPSK Transmitter and Recelver 4.34.1 Transmitter | Generator of DPSK Signal Fig. 33.1 shows the scheme to generate DPSK signal. Input coquerce oo be 0) = RP cos(2ntgt) jo = PF costa) DPSK sigral ae) Fig, 3.3.1 Block diagram of DPSK generate or transmitter Operation and waveform of transmitter ‘The input sequence is d(). Output sequence is b(t) and b(t-T,) is the previous output delayed by one bit period. Depending upon values of d(t) andb(t~T,), exclusive OR gate generates the output sequence b (). Table 23.1 shows the truth table of this operation. ag be-T) ow o(-vy o(-tv) oy ov) avy sv) 10¥) O(-1¥) 10v) 11V) 10) ov) Table 3.3.1 Truth table of exclusive OR gate An arbitrary sequence 4 () is taken. Depending on this sequence, b() andb(t-T;) are found. These waveforms are shown in Fig, 3.32. The above table 3.3.1 is used to derive the levels of these waveforms.Digital Communication 3-45 Passband Data Transmission Fig. 3.3.2 DPSK waveforms From the waveforms of Fig. 33.2 it is clear that b(t Tj) is the delayed version of b(®) by one bit period T,. The exclusive OR operation is salisfied in any interval ie. in any interval b(®) is given as, b®) = d(\@b(t-T,) (33.1) While drawing the waveforms the value of b(t~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(() changes level at the beginning of each interval in which 4 (i) =1 and it does not changes level when d (#)=0. Observe that d(3) =1, hence level of # (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(t)=0. Hence b(t) is not changed in these intervals. 2. When d(t)=0, b() =b¢-T,) and When d(t)=1, 6) =5¢-T,) 3, In interval no.1. we has assumed b(t~7],)=0 and we obtained the waveform as shown in Fig. 33.2. If we assume b(t-T,)=1 in interval no. 1, then the waveform of b(t) will be inverted. But still b() changes the level at the beginning each interval in which d(t)=1. 4. The sequence b(t) modulates sinusoidal carrier.Digital Communication 3-18 Passband Data Transmission 5. When h(i) changes the level, phase of the carrier is changed. Since b()) changes its level only if d'()=1; It shows that phase of the carrier is changed only if dah In BPSK phase of the carrier changes on both the symbol ‘1’ and ‘0’. Whereas in DPSK phase of the carrier changes orly on symbol ’1'. This is the main difeerence between BPSK and DPSK. J 6. Always bvo successive bits of d() are checked for any change of level. Hence one symbol has two bits. Symbol duration (T) = Duration of two bits (2T,) ie. T=2, ~ (8.32) As shown in Fig. 3.3.1, the sequence b() is applied to a balanced modulator. The balanced modulator is also supplied with a carrier J2P cos (2x fy t). The modulator output is, 5) = DONE es (2nfy0) G33) V2B cos (2x fot) . (3.34) The above equation gives DPSK signal. Fig. 3.3.2 shows this DPSK waveforms. As shown in the waveforms the phase changes only when d (i) =1. 9.3.4.2 DPSK Receiver Fig, 333 shows the method to recover the binary sequence from DPSK signal. Fig. 333 (a) and (b) are equivalent to each other. Fig. 33.3(b) represents DPSK receiver using correlator. Fig. 33.3(a) shows multiplier and integrators separately. Operation of Receiver 1, Phase shift in received signal : During the transmission, the DPSK signal undergoes some phase shift 8. Therefore the signal received at the input of the receiver is, Received signal = b(t) V2P cvs (2n fy t +8) wo» 3.35) 2. Multiplier output : This signal is multiplied with its delayed version by one bit. Therefore the output of the multiplier is, Multiplier output = # (0) b(t ~7),) (2P) cos (2n fot +0) cos [2m fg (t-T))+O] —~. (3.3.6) We know that, cos (A) cos (B) = cos (A ~B) + cos (A +B)]Digital Communication 3-17 Passband Data Transmission b(t b(Ty) (QP) cos(enfet+ 6) cosfeat(t-Te)+ Ol C Integrator yet > A Bit synchronizer Correiator }—__! b4t-T,) VaP cos[2efot-T,)+0] (>) Fig. 3.3.3 (a) DPSK receiver (b) Equivalent diagram of DPSK receiver using correlator Here A= 2xfyt+@ and nf (t= Th) +0 :. Multiplier output = (004-75) [ees 2nfo Ty e041 +m} G37) l fo is the carrier frequency and 7, is one bit period. T,, contains integral number of cycles of fo. We know that, 1 fo = Ty If 7, contains ‘n’ cycles of fo then we can write, fo fe > hae fol) =" + (33.8)igital Communication 3-18 Passband Data Transmission Putting fy T = 1 in first cosine term in equation 3.3.7 we get Multiplier output = 6()b(¢-T,)P fe 2nnt wtf (« - %) +a} Since cos 2xn=1, the above equation will be, Multiplier output = b()b¢-T,) P+b@b¢ Tay Pex tf (+ - 3) »| 83.9) 3. Integrator : The above signal is given to the integrator. In the k** bit interval, the integrator output can be written as, kT 5o(kTp) = B(RT,O[R-DT]P fat DT kt 1b + var OLK-OT]P J cx [snsa(+- Tb) 20a kT The integration of the second term will be zero since it is integration of carrier over one bit duration. The carrier has integral number of cycles over one bit period hence integration is zero. Therefore we can write, Sy(kTy) = b(KT,)b[(K-1)T,] P[kT, ~(k-1)T,] b(KT,)O[K-)T] PT, =» B.3.10) Here know that PT, =E, ; ie. energy of one bit. The product b(kT)b [(k-1)T}] decides the sign of P Ty. The transmitted data bit d(t) can be verified easily from product b(KT4) b[K-1) Tp]. We know from Fig. 33.2 when b()=b(¢-T)),d (=O. That is if both are +1V or -1V then b()b(!-T,) =1. Altemately we can write, If o@b(t-7,) =v then di) = 0 We know that b()=b(=T;) then d(t)=1. That is b()=-1V,b((-7,)=+1V and vice versa, Therefore b({)b(I-T,)=—1. Aliernately we can write, If b@bE-T,) = =v, then d() = 1 4, Decision device : The decision device is shown in Fig, 33.3 (b). We know that, $5 KT) = b(RKT))b[K-T] PT, from equation 3.3.10Digital Communication 3-19 Passband Data Transmission {[=PTi,then d()=1 and +PTp, then d(t)=0 i 59 (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 () occurs only if input bit is at level “I’, No change occurs if input bit is at level '0. ce one previous bit is always used to define the phase shift in next bit, the symbol can be said fo have Heo bits, Therefore one symtol duration (T) is equivalent to to bits duration (27). Symbol duration T = 27; Bandwidth is given as, BW sie 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 each bi independent.Digital Communication 3-20 Passband Data Transmission 3) Noise interference in DPSK is more. In DPSK, previous bit is used to detect next bit. Therefore if error is present in previous bit, detection of next be can also go wrong. Thus error is created in next bit also. Thus there is tendency of appearing errors in pairs in DPSK. wm> Example 3.3.1: The bit stream 1011100011 is to be transmitted using DPSK. Determine the encoded sequence and transmittd phase sequence Solution : Fig. 3.3.4 shows the encoded bit stream b(t) and the transmitted phase. The input bit stream is represented as dil). The encoding waveforms are shown as per the DPSK generator of Fig. 3.3.1 ond Table 33.1 (t~T}) is the encoded sequence delayed by one bit period. sequence b(t Lf t Transmitted to 4 t hase sequence 00} nb 0 Ob et Lott Ionia bn I 1 Fig. 3.3.4 DPSK waveforms In the above figure observe that the delayed output sequence bit -T,) is assumed intially. The encaded sequence bi) is given as, &) = a) @b¢-T,) From equation 3.3.1 | | |Digital Communication 3-21 Passband Data Transmission In the Fig. 3.3.1 observe that the transmitted signal is given as, s(t) = b(t)V2P cos(2f,6) +V2P cos(2nf,t) V2P cos(2nj,t) when b(t)=1 {. V2P cos(2nfyt) when Wb ‘The above equations can also be written as, st) = { V2P cos(2nft+0) when U(t)=1 ie. st) = - VPP cost2xf,t+m when Bit) =0 ‘The transmitted phase sequence is shown in Fig. 3.34 as per the above equation. Review Questions 1. With the help of Block diagram, waveforms and expressions explain the operation of DPSK transmitter and receiver. 2. What ave the advantages and disadvantages of DPSK ? What is the bandwidth requirement of DPSK ? 3.4 Quadrature Phase Shift Keying May/ora: 2008 Principle + In communication systems we know that there are two main resources, i.e. transmission power and the channel bandwidth. The channel bandwidth depends upon the bit rate or signalling rate f,. In digital bandpass transmission, a carrier is used for transmission, This carrier is transmitted over a channel. «If two or more bits are combined in some symbols, then the signalling rate is reduced. Therefore the frequency of the carrier required is also reduced. This reduces the transmission channel bandwidth, Thus because of grouping of bits in symbols, the transmission channel bandwidth is reduced. ‘+ In quadrature phase shift keying, two successive bits in the data sequence are grouped together. This reduces the bits rate of signalling rate (ie. f,) and hence reduces the bandwidth of the channel. «In BPSK we know that when symbol changes the level, the phase of the carrier is changed by 180° Since there were only two symbols in BPSK, the phase shift occurs in two levels only. + In QPSK two successive bits are combined. This combination of two bits forms four distinct symbols. When the symbol is changed to next symbol the