CHAPTER - I LAPLACE TRANSFORM

1.1 INTRODUCTION The analysis and design of many physical systems is based upon the solution of an ordinary linear differential equation with constant coefficients. This is, in fact, an idealization of the actual process. Nevertheless, in a defined operating region, many systems can be described by an ordinary linear differential equation. The procedure is illustrated in Fig. 1.1.

In general, physical laws are applied to the physical system to obtain its mathematical description. Since the operating region of many systems covers a small range, this mathematical description can, therefore, be linearized to give ordinary linear differential equations. Later, initial conditions, if any, can be added and the solution obtained to give information for design or analysis. The solution of such equations can either be obtained by known methods of ordinary linear differential equations or by transform methods. The former method of substituting an assumed solution in the differential equation and then finding the values of constants is quite laborious. In the latter, the use of the Laplace transform changes the differential equation into an algebraic equation. This transformation changes the differential equation with time as the independent variable into an algebraic equation with s as the independent variable. The initial conditions are added automatically during the process of transformation and the time solution is found by inverting the transformed equations. This method is also applicable to partial differential equations.

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1.2 DEFINITION Let f(t) be a function of t with the following properties: 1. f(t) is identically zero for t < 0 2. f(t) is continuous from the right at t = 0 Mathematically, 1. f(t) = 0 for t < 0 2. limt→0 f(t) = f (0) for t > 0 Then the Laplace transform of f (t), denoted by L[f (t)], is defined as Z ∞ e−st f(t) dt L[f(t)] = F (s) =
0

(1.1)

where s = σ + jω is a complex number. The Laplace transform of Eq. (1.1) shall exist if and only if the integral converges for some values of σ. 1.3 SUFFICIENT CONDITIONS FOR EXISTENCE OF LAPLACE TRANSFORMS THEOREM 1.1: If f(t) is sectionally continuous in every finite interval 0 < t < N and of exponential order for t > N , then its Laplace transform F (s) exists. PROOF: In order to prove the theorem, let us define the function of exponential order. Definition 1.1 If real constants M > 0 and γ exist such that for all t > N, |e−γt f(t)| < M or |f (t)| < M eγt we say that f(t) is a function of exponential order γ as t → ∞. Now, we have for any positive number N, Z
∞ 0

(1.2)

e −st f(t)dt =

Z

N

e− stf (t)dt +
0

Z

e −st f(t)dt

(1.3)

N

Since f (t) is sectionally continuous in every finite interval 0 ≤ t ≤ N, the first integral on the right exists. Also, the second integral on the right exists, since f (t) is of exponential order for t > N. To see this, let us observe that ¯Z ∞ ¯ Z ∞ ¯ ¯ ¯ e− stf (t) dt¯ ≤ |e−st f(t)| dt (1.4) ¯ ¯ N ZN∞ ≤ e− σt |f (t)| dt, s = σ + jω 0 Z ∞ < e−σ t Me γt dt
0

=

M ,σ > γ σ−γ
2

(1.5)

Thus the Laplace transform exists for Re{s} > γ. If this sufficient condition is not satisfied, then the Laplace transform for f(t) may or may not exist. 1.4 LAPLACE TRANSFORM OF FUNCTIONS We consider here some elementary functions which are used in engineering problems. Example 1.1 Unit Impulse Function

The unit impulse function or delta function is denoted by the symbol δ(t) (Fig. 1.2), and is defined by the relationships δ(t) = 0, t 6= 0 and Z
b

δ(t) dt =
a

n

1, a ≤ 0 ≤ b 0, otherwise

Hence, δ(t − t0 ) = 0, t 6= t0 It has the property that Z
b

a

δ(t − t0 ) dt =

Which implies that δ(t) has infinite magnitude at t = t0 . In addition, for any function f(t) Z
b

n

1, for a ≤ t0 ≤ b 0, otherwise

(1.6)

a

f (t)δ(t − t0 ) dt =

n

f (t0 ), for a ≤ t0 ≤ b 0, otherwise

(1.7)

δ(t) is transformable for every s, since by Eq. (1.7) Z Hence, L[δ(t)] = 1 Example 1.2 Define u(t) = Unit Step Function n 1, t ≥ 0 0, t < 0 (1.9)
∞ 0

e− stδ(t) dt = 1

(1.8)

The graph of the function is shown in Fig. 1.3 Then, U(s) = Z
∞ 0

¯∞ ¯ 1 1 1.e−st dt = − e −st ¯ = ¯ s s 0
3

Hence, 1 L [u(t)] = , Re{s} > 0 s Example 1.3 Unit Ramp Function (1.10)

The function is shown in Fig. 1.4 and defined as f (t) = t, for t ≥ 0 Then, F (s) = Z
∞ 0

te−st dt te−st e −st + 2 s s ¸∞
0

=− = Therefore, L [f(t)] = 1 s2 Example 1.4 Exponential Function

1 , Re{s} > 0 s2

(1.11) (1.12)

The graph of the function is shown in Fig. 1.5. Let, Z
∞ 0 ∞ 0

f(t) = e−at , for t ≥ 0 e− at e− st dt e− (s+a)t dt (1.14) (1.13)

F (s) = =

Z

=− Example 1.5

Sinusoidal Function

¯∞ 1 − (s+a)t ¯ ¯ = 1 , Re{s} > −a e ¯ s+a s +a 0

The function f (t) = sin(at) can be written as f (t) = sin(at) = Then, F (s) = ∙Z ∞ ¸ 1 (ejat − e− jat )e− st dt 2j 0 ∙Z ∞ ¸ Z ∞ 1 e−(− ja+s) t dt − e−( ja+s )t dt = 2j 0 ∙ ¸ 0 1 1 a 1 , Re{s} > 0 − = 2 = 2j s − ja s + ja s + a2
4

ejat − e− jat 2j

(1.15)

.

6 Find the Laplace transform of f(t) if f (t) = n 4.) We reduce this equation by integrating ˙ dt by parts Z ∞ 0 L{y(t)} = y(t)e−st |0 − ˙ ∞ y(t). we will frequently use y to represent dy .Example 1. ∀ s s 0 Z ∞ (1. We wish to find Ly(t) = ˙ Z ∞ 0 y(t)e− st dt ˙ (1.7 Let f(t) = t2 Then integrating by parts. Re{s} > 0 s The reader should verify that lim t2 e−st = 0 t→∞ (1.e−st dt = 0 0 ¯ − st ¯2 ¯ 4e ¯ ¯ =¯ ¯ −s ¯ 0 4 = [1 − e− 2s ] .16) Example 1.18) (In this text.(−s. 0 ≤ t < 2 0.17) 1.5 LAPLACE TRANSFORM OF DERIVATIVES Let us now proceed to find the Laplace transform of derivatives. L[y(t)] = Y (s). Define.e −st ) dt (1. we have Z∞ F (s) = t2 e− st dt 0 ¯∞ ¯∞ ¯∞ ¯ ¯ ¯ 1 2 2 = − e−st t2 ¯ − 2 e −st t¯ − 3 e−st ¯ ¯ ¯ ¯ s s S 0 0 0 2 = 3 . t ≥ 2 By definition L[f(t)] = f (t)e−st dt Z ∞ Z2 e− st dt + 0.19) = − y(0) + sY (s) The only restriction is that y(t) must be such that 6 .

8 Consider the circuit of Fig.20) dz dz d 2 z − st e dt = sL( ) − (0) dt2 dt dt (1.23) where L0 is the inductance.2.19) by letting y(t) = Thus d 2z = dt2 Also L( dz ) = sZ(s) − z(0) dt Combining the above equations L ∙ ¸ d2 z dz = s 2 Z(s) − sz(0) − (0) dt2 dt (1.21) Some of such results are given in Table 1.22) Z ∞ 0 dz dt (1. 1. Transforming the equation (1.6 LAPLACE TRANSFORM APPLIED TO ORDINARY DIFFERENTIAL EQUATIONS Example 1.23) 7 .6 The equation describing the electric current i(t) is L0 di + Ri = E dt (1.lim y(t)e−st = 0 t→∞ The Laplace transforms of higher derivatives are easily deduced from Eq. Other properties are the subject of Chapter IV. R is the resistance and E is a constant voltage source. (1. 1.

we get i(t) = E (1 − e−R/L0t ).26) E s E s(sL0 + R) E L0 s(s + R L0 ) Example 1. then dt 8 . then L0 sI (s) + RI(s) = or (sL0 + R)I(s) = I(s) = = separating by partial fraction method # " E 1 1 I(s) = − R R s s + L0 Inverting I(s) into time domain.25) If we assume that initial current i(0) in the inductor is zero.28) [s2 Y (s) − sy(0) − If y(0) = 0 and dy (0) = 0. R t≥0 (1.¸ ∙ di L L0 + Ri = L[E] dt L0 L ∙ ¸ di + RL[i] = L[E] dt E s (1.9 Consider the differential equation dy d2 y + 4 + 3y = sin(t) dt2 dt ∙ ¸ d2 y dy + 4 + 3y = L[sin t] dt2 dt 1 dy (0) + 4sY (s) − 4y(0) + 3Y (s)] = 2 dt s +1 L (1.27) E s (1.24) L0 [sI(s) − i(0)] + RI(s) = (1.

30) 9 .Y (s) = 1 (s2 + 1)(s2 + 4s + 3) 1 = (s + 1)(s + 3)(s2 + 1) (1.4636).31) in the time domain can be obtained and is given below: 1 1 1 y(t) = e−t − e −3t − √ cos(t + 0.29) The solution of Eq. t ≥ 0 4 20 20 (1. (1.

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dt2 2 d y dt3 3 (0) = 0. (0) = 1. (0) = 0.1 Find the Laplace transforms of y(t) in the following differential equations.2 Find the Laplace transforms of the following functions of time. y(0) = 1 1. a) b) c) d2y dt2 d y dt4 d y dt3 3 4 − dy dt − 2y = 0. dy dt dy dt (0) = 0. y(0) = 1. y(0) = 1 + 7 d y + 12 dy = (1 + t)e−3t . a) f(t) = tn b) f(t) = sin (ωt) c) f (t) = te− at d) f(t) = e− at sin(ωt) e) f (t) = t2 sin(t) 2 11 .PROBLEMS 1. dt2 dt (0) = 0. 2 dy dt (0) = 2 d 2y dt2 d y dt2 2 − 2 d y + y = 0.

2.2 1 s(M s2 + f s + k) (2. Transforming Eq. the resultant of the forces is zero. For rotational systems.2 By Newton’s Law: dθ d2θ J 2 = −B + T − kθ dt dt J dθ d2 θ +B + kθ = T dt2 dt 12 (2. we obtain 1 s 2 M X(s) + sfX(s) + kX(s) = s so that X (s) = Example 2. (2. K the stiffness coefficient.2) and assuming that all initial conditions are zero and F(t) is a unit step function. If the forces on the body are balanced.2) M Where M is the mass of body.2 MECHANICAL SYSTEMS The differential equations for mechanical systems are written by using Newton’s Law which states that for a translational system the sum of the forces acting on a body is equal to the mass times the linear acceleration of the body.1 Translational System Consider the translational mechanical system in Fig. These transformed equations can then be analyzed by the s-plane analysis techniques discussed later. i.3) Linear Rotational System Consider the rotational system of Fig. the law states that the sum of torques acting on a body is equal to the moment of inertia times the angular acceleration of the body. Example 2.1) (2.II APPLICATIONS TO PHYSICAL SYSTEMS 2. the force in the positive direction is equal to the force in negative direction. 2.1 INTRODUCTION The aim of this chapter is to give an appreciation of the equations of linear physical systems and their formulation in the Laplace domain.CHAPTER .e. hence the acceleration is zero.4) . 2. In this case. f the friction coefficient. the mass will not move.1 whose differential equation for applied force F(t) is: M or d2 y dx + F (t) = −kx − f dt2 dt d2 x dx + kx = F (t) +f dt2 dt (2. and x denotes displacement.

2.4 which is made up of an inductive coil.4) and assuming that θ(0) = 0. (2.3 T (s) Js2 + Bs + k (2. The analysis of electrical circuits is based on two fundamental laws: Kirchoff’s voltage law and Kirchoff ’s current law.7) 2. across the capacitor: eC (t) = C 0 i(u)du + e C (0) where the voltage e c(0) is caused by charges already on the capacitor at time t = 0.3 whose differential equation is given below: M1 dx1 d2 x1 =0 + (k1 + k 2)x1 − k2 x2 + f dt2 dt 2 d x2 M2 2 + k2 x2 − k 2x1 = F (t) dt Transforming M1 s2 X1 (s) + (k1 + k2 )X1 (s) − k2 X2 (s) = 0 M2 s 2 X2 (s) + k 2X2 (s) − k2 X1 (s) = F (s) ¸∙ (2. As the current i(t) flows through the elements of the circuit. B the friction coefficient. a resistor. A time varying voltage is applied across terminals 1 and 2. T the applied torque. Both these laws apply to instantaneous values of voltages and currents. The second law states that the sum of all currents entering a node is zero.Where J is the moment of inertia.3 ELECTRIC CIRCUITS Consider the simple electric circuit shown in Fig. and θ the angular displacement. This excitation produces the current i(t). and the voltage e2 (t) across terminals 3 and 4.5) Coupled Translational System Consider Fig. Applying Kirchoff’s voltage law to the closed path: 13 . ˙ Transforming Eq. across the inductor: eL (t) = L di(t) dt 2. and a capacitor. it causes a voltage drop that is opposed to the direction of current flow. 2. The magnitudes of these voltage drops are: 1.6) can now be written in matrix form as follows: ∙ −k 2 M1 s2 + f s + k 1 + k2 −k 2 M2 s 2 + k2 ¸ ∙ ¸ 0 X1 (s) = F (s) X2 (s) (2. we obtain ˜ θ(s) = Example 2. The first law states that the sum of all voltages around any closed path of a circuit is zero.6) Eq. θ (0) = 0 . across the resistor: eR (t) = Ri(t) Rt 1 3. (2.

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6.8) and (2. We derive its mesh equations in Laplace domain. The voltage across terminals 3 and 4 is equal to the voltage across the capacitor.8) L dt C 0 This is an integro-differential equation defining the unknown function i(t). 2. 2.11) The above set of equations can now be solved for unknown variables. Substitution of the voltage drops yields Z 1 t di(t) + Ri(t) + i(u) du + eC (0) = e1 (t) (2.4 Consider the passive network (Fig. The circuits for mesh and nodes are given in Figs. a set of nodal equations can be written from Fig. Example 2. namely 1 C Z t i(u) du + e C (0) = e 2 (t) 0 (2. 2.9) can now be Laplace transformed. voltages which oppose the clockwise current flow have a negative sign. The mesh equations for the network can be written as follows: µ ¶ 1 1 V (s) = R1 + I (s) − I (s) C1 s 1 C1s 2 0=− 1 I1 (s) + C1 s ¶ 1 1 1 + + sL1 I2 (s) − I3 (s) C1 s C2 s C2 s ¶ µ 1 1 I2 (s) + R2 + I3 (s) 0=− C2 s C2 s µ (2.6) whose equations have to be derived by mesh and nodal methods. 2.7.10) Similarly.5 An equivalent circuit for an active device is shown in Fig.9) Equations (2. Example 2. The following mesh equations for the circuit can easily be derived with the help of Kirchoff ’s voltage law.5 or 2. we wish to solve the system for e 2 (t).5 and 2.e1 (t) − eL (t) − e R(t) − eC (t) = 0 Here.6 as given below: V (s) 1 1 V (s) = 1 + sC 1 V1 (s) + V (s) − V (s) R1 R1 sL1 1 sL1 2 0=− 1 1 1 V1 (s) + sC2 V2 (s) + V2 (s) + V2 (s) sL1 sL1 R2 (2. However. Vs (s) = (Rs + Rg )I1 (s) − Rg I4 (s) 15 .

.

from Fig.−µV g (s) = (Rp + RL )I2 (s) − RL I3 (s) 0 = −RL I2 (s) + (RL + R2 + 1 )I (s) − K R2 I4 (s) sC 3 (2. (2. 2.15) µ s (2.4 A SIMPLE THERMAL SYSTEM Consider a large bath whose water temperature is at µ degrees and T is its lag constant. A thermometer indicating θ degrees is immersed in the bath.8).5 A SIMPLE HYDRAULIC SYSTEM Consider the system shown in Fig. Assume that q0 the volumetric flow rate through the resistance is related to the head h by the linear relationship: 17 .9 which consists of a tank of cross sectional area A to which is attached a flow resistance R such as a pipe. for example ⎡ ⎤⎡ ⎤ ⎡ ⎤ Rs + R g 0 0 −Rg V I (s) µ 0 Rp + RL −RL ⎢ ⎥ ⎢ I1 (s) ⎥ ⎢ 0s ⎥ SC gk ⎢ ⎥⎣ 2 ⎦ = ⎣ ⎦ (2. 2. The equation is 1 ˙ θ(t) = (µ − θ(t)) T Transforming Eq.16) s(s + T1) + θ(0) 1 s+ T (2. The unknown variables can be determined by using the Cramer’s rule.13) The matrix equation can be written as follows. Newton’s Law of Cooling states that the rate of change of temperature is proportional to the difference between the bath temperature and the measured temperature (Fig.15) T [s˜ − θ(0)] + ˜ θ(s) θ(s) = ˜ = θ(s) µ T (2.7.14) 1 ⎣ 0 −RL RL + R2 + SC −KR2 ⎦ I3 (s) 0 1 0 I4 (s) −Rg 0 −KR2 Rg + K2 R + SC gk 2. Vg (s) = 1 I (s) sCgk 4 (2. 2.12) ∙ ¸ 1 I4 (s) 0 = −Rg I1(s) − K R2 I3 (s) + Rg + KR2 + sC gk Also.17) 2.

.

Each group is aggregated into one function. 19 . 2.20) to eliminate q0 (t) gives ρq(t) − ρq0 (t) = q− h dh =A R dt (2. The rate of flow of demand. d (ρAh(t)) dt dh q(t) − q0 (t) = A dt Combining Eqs. sells to the consumer. They are: 1. The mass balance equation around the tank is: Mass flow in .10 shows a diagram for the market of a single commodity.23) (2. τ = AR Equations for hydraulic systems with more than one tank can easily be obtained by following the above method. d(t). (2.21) Q(s) = H(s) + AsH(s) R 1 + As)H (s) R RQ(s) 1 + τs (2. measured in units of commodity per unit time. purchases from the supplier. The variables of the market are assumed to be continuous functions of time. The stock level.q0 = h R (2. 2. The merchant sets the price.18) Liquid of constant density ρ enters the tank with volumetric flow q(t).22) Q(s) = ( H(s) = where. s(t). Three groups are involved in the market: the suppliers. q(t). measured in units of commodity.19) (2. 2. and maintains a stock of the commodity. and the consumers.20) (2. The rate of flow of supply.mass flow out = rate of accumulation of mass in the tank. Determine the transfer function which relates head to flow. measured in units of commodity per unit time.18) and (2. 3. (2. So that.21) Assuming initial conditions as zero and transforming Eq.6 A MODEL OF A SINGLE COMMODITY MARKET Fig. the merchants.

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p(t) = A[qi − q(t)] ˙ (2. In modelling this market. The price per unit of commodity. Negative values of q(t) will be interpreted as the amount of the commodity which has been sold. C may be either positive or negative. A more elaborate model considers the anticipatory nature of the supplier as well as the consumer reaction.4. This function is shown in Fig. the consumer may purchase more than he needs because prices are rising and because he wishes to buy before the prices are too high. (2. The second assumption is that the merchant sets the price p(t) at each instant of time. any access of supply over demand goes into stock. (2. assumptions need to be made about the relationship of the above variables. Thus.29) .25) and differentiating the resulting equation. First of all. s(t) − d(t) = B[p(t) − Pe ] (2. (2. (2.26) becomes: s(t) − d(t) = B[p(t) − Pe ] + C p(t) ˙ (2. The third assumption is that both supply and demand depend on the price. the supply s(t) increases linearly with price and the demand d(t) decreases linearly with price. p(t).2.24) in Eq. we obtain d2 pt = −A[s(t) − d(t)] dt2 Substitution of Eq. If both of these effects are present.28) (2.25) Where A is a positive constant. Eq. then C > 0.27) yields the following second order differential equation d2 p(t) dp(t) + ABp(t) = ABPe + AC dt2 dt which can be Laplace transformed.26) Where B is a positive constant and Pe is the equilibrium price for which supply equals demand.24) An excess of demand over supply is filled from stock. then C < 0. This provides for an increase in supply or demand proportional to the rate of change of price. but not delivered because of shortage. 21 (2. In the simplest model.11. Substituting Eq. Mathematically. For example. so as to make the rate of increase of p(t) proportional to the amount by which the actual stock q(t) deviates from an ideal stock level qi . This means that: q(t) = q(0) + Z t 0 [s(u) − d(u)] du (2. With such a term added. On the other hand.27) The value of C may be positive or negative. if it is assumed that the supplier increases his production because prices are on the rise and because he wishes to benefit from a greater profit margin. the excess of supply over demand is linearly increasing function of p.

3 For the circuit in Fig. 2. Now let x(t) be an arbitrary function of time.14(c). 2. the leading car has a higher (lower) speed.12.31) 22 . Find ic(t). The leading car has the position x(t).14(a). for the initially inert circuit and write the equation in Laplace transform. The actual acceleration is the result of both effects. (2.1 Write the Laplace transformed differential equations for the three mechanical systems in Fig. Both x(t) and y(t) are measured from the same reference point and both increase with time.7 A MODEL FOR CAR FOLLOWING Consider the position of two cars moving on a single lane road.30) = A[x(t) − y(t) − d] + B dt2 dt dt Where A and B are positive constants. the driver will accelerate (decelerate) in proportion to the deviation from d. n t. 2. Separation of x(t) and y(t) in this equation yields: dx(t) dy(t) d2 y(t) + Ay(t) = Ax(t) + B − Ad +B dt2 dt dt Define. Find the transfer function E2 (s)/E1 (s).2. PROBLEMS 2. the driver will accelerate(decelerate) in proportion to d2 y(t) the difference in speed. the current through the capacitor.31) is a typical second order differential equation. It follows that x(t) > y(t).5 In the circuit shown in Fig. let e(t) = Cos(ωt). 2. 1 0. and that x(t) = y(t) implies the collision of the cars. 2. Find iL (t) and write the Lalace transformed equations.13. where passing is not possible.14 (b). 2. the switch is closed at t = 0. for 0 < t. for t ≥ 1 (2. Moreover. that is dt2 ¸ ∙ d 2 y(t) dx(t) dy(t) − (2. 2.2 Write the Laplace transformed differential equation for the electric circuit in Fig. If the distance increases (decreases). and the second car has the position y(t). let the circuit be initially inert and let e(t) = Find the charge q(t) on the capacitor.4 For the circuit in Fig. f(t) = Ax(t) + Bx(t) − Ad ˙ Now Eq. 2. 2. Assume that the initial distance is x(0)−y(0) = d and that the driver of the second car attempts to hold the distance d at all times.

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25 . 0 An example of a null function is n(t) = n 1. then f (t) and g(t) are identical functions. for all t > 0.. 2... We can. 0.. . the behaviour of a physical system is determined via following steps: 1. reached the stage where we could obtain the desired variables. Obtain the initial conditions. 1. Derive the ordinary differential equations describing the system. Inversion integral The following discussion is limited to systems which have the following form: Y (s) = a n sn + an− 1 sn−1 + . 5. A null function has the property Z t n(τ)dτ = 0. we consider the problem (step 5.1) The coefficients ai and bi are real and m. 2. which can take care of a wide range of problems. n 0. . above) of finding the original function f (t) from its image function F (s). . In this chapter. 4. we have to invert the Laplace transform. There are three methods to obtain the time domain solutions 1. We have. L−1 [L[f (t)]] = f (t) The uniqueness means that if two functions f(t) and g(t) have the same transform F (s). In order that the transform calculus be useful. There is a theorem which states that two functions f(t) and g(t) that have the same Laplace transform can differ only by a null function n(t). that is. Manipulate the algebraic equations and solve for the desired dependent variable. . for t = 1. 2.CHAPTER . . therefore.. we must require uniqueness. . Laplace transform tables 2. + b1 s + b0 (3. 3. . The most obvious method to find the inverse is to use Laplace transform tables. so far.. 3.III THE INVERSE LAPLACE TRANSFORM 3. Partial fractions 3.1 INTRODUCTION In the Laplace transform method. Laplace transform the differential equations including the initial conditions. otherwise Null functions are highly artificial functions and are of no significance in applications. Find the inverse Laplace transform to obtain the solution in time domain. In order to obtain the solution in time domain. + a1 s + a 0 bm sm + bm− 1 sm− 1 + .. say that the inverse Laplace transform of F (s) is essentially unique. .

As an example. it should be first divided. Then: A= Similarly. To evaluate A. (3.5) holds for all values of s we may let s = −1 and solve for A.4) The partial fraction expansion shall be complete when we have evaluated the values of the constants A. and C. so that s 2 + 12s + 14 B(s + 1) C(s + 1) = A+ + (s + 1)(s + 2)(s + 4) s+2 s+4 (s + 1) (3. consider the following: Example 3.5) Since Eq.2 PARTIAL FRACTION METHOD The rational fraction in Eq. B. Step 2 Factor the polynomial in the denominator of Y1 (s).2) Y (s) = 1 + (3. multiply both sides by (s + 1). (3. let s = −2 B= and. so that s 3 + 7s2 + 14s + 8 = (s + 1)(s + 2)(s + 4) Step 3 Y1 (s) = A B C s2 + 12s + 14 = + + s 3 + 7s2 + 14s + 8 s+ 1 s+ 2 s+4 (3.1 Consider Y (s) = Step 1 Eq.3.2) cannot be expanded into partial fractions because the degree of numerator and denominator is equal and. therefore. so that s2 + 12s + 14 = 1 + Y1 (s) s 3 + 7s2 + 14s + 8 s 3 + 8s2 + 26s + 22 s3 + 7s 2 + 14s + 8 (3.1) can be reduced to a sum of simpler terms each of whose inverse Laplace transform is available from the tables. let s = −4 26 (−1)2 + 12(−1) + 14 3 = =1 (−1 + 2)(−1 + 4) 3 −6 (−2) + 12(−2) + 14 = =3 (−2 + 1)(−2 + 4) −2 2 .3) Now Y1 (s) can be expanded into partial fractions. (3.

9) (3.C= so that Y1 (s) = and −18 (−4)2 + 12(−4) + 14 = = −3 (−4 + 1)(−4 + 2) 6 1 3 3 s 2 + 12s + 14 = + − s + 7s 2 + 14s + 8 s+1 s+2 s+4 3 (3. t ≥ 0 Example 3.2 Consider another example Y (s) = s2 + 3s + 1 s(s + 1)3 (3.10) The two constants A and B3 can be determined as before ¯ s(s2 + 3s + 1) ¯ ¯ =1 3 ¯ s(s + 1) s=0 ¯ (s + 1)3 (s 2 + 3s + 1) ¯ ¯ =1 B3 = ¯ s(s + 1)3 A= s=−1 The other coefficients are found by differentiating.6) Y (s) = 1 + 1 3 3 + − s+1 s+2 s+4 (3.8) Because of third order pole at s = −1.7) whose inverse transform is (from Tables) y(t) = δ(t) + e− t + 3e −2t − 3e− 4t . To find B2 . we write the expansion Y (s) = B1 B2 A B3 s2 + 3s + 1 + = + + s(s + 1)3 s s + 1 (s + 1)2 (s + 1)3 (3. we differentiate once # " 3 i d (s + 1) (s2 + 3s + 1) d h 3 (s + 1) Y (s) = 3 ds ds s(s + 1) ∙ 2 ¸ d s + 3s + 1 = ds s ¯ s(2s + 3) − (s 2 + 3s + 1) ¯ ¯ = ¯ s2 s=−1 ¯ 2 ¯ s − 1¯ = =0 s2 ¯ s=−1 27 .

(3. And ∙ A ¸ ∙ ¸ d2 d2 d2 d2 (s + 1)3 + B1 2 [(s + 1)2 ] + B2 2 (s + 1) + 2 B3 = 2B1 ds2 s ds ds ds s=−1 so that B1 = −1 Therefore Y (s) = and from transform tables y(t) = 1 − e− t + Example 3.13) t2 −t e .16-3. Y 1(s) = α = 2 and ω = 2 Using Eq.11) The first constant A is easily found ¯ ¯ −1 s+1 ¯ = (s + 2)2 + 4 ¯s=− 2 4 s+ 1 s+ 2 A= To evaluate B and C . 2 t≥0 (3.12) 1 1 1 − + s s + 1 (s + 1)3 (3.and ∙ ¸ ∙ ¸ d d (s + 1)3 d d 2 + B1 (s + 1) + B2 (s + 1) + B3 A = B2 ds s ds ds ds s=− 1 ∙ ¸ i d s2 − 1 d2 h (s + 1)3 Y (s) = ds2 ds s2 ¯ s2 (2s) − (s2 − 1)2s ¯ ¯ = −2 ¯ s4 s=−1 so that. (3. B2 = 0 For B1 . we follow the method outlined in Eqs.20) In that notation.14) .3 Consider another system with complex poles Y (s) = s+1 (s + 2){(s + 2)2 + 22 )} A Bs + C = + s + 2 (s + 2)2 + 4 (3.20) 28 (3.

Since A[s + (α − jω)] is complex conjugate of B[s + (α + jω)] we can write 2ReA[s + (α − jω)] 2 (s + α) + ω 2 Y (s) = Evaluating the coefficient A. = s + (α + jω) s + (α − j ω) (s + α)2 + ω2 A[s + (α − jω)] + B[s + (α + j ω)] (s + α) + ω2 2 (3.18) But since Y (s) has real coefficients...17) The first two terms can be combined as (3.19) A= ¯ (s + (α + jω))Y1 (s) ¯ ¯ 2 (s + α) + ω2 ¯s=− α−jω Y1 (−α − j ω) Y (−α − jω) = = 1 −α − jω + α − jω −2jω 29 . Y (s) = Y1 (s) B A + + .15) This function is inverted directly from the tables as follows ∙ ¸ ∙ ¸ ∙ ¸ 1 −1 s+2 1 4 + L−1 + L− 1 2 2 4(s + 2) 4 4 (s + 2) + 22 (s + 2) + 22 y(t) = L− 1 [Y (s)] = L− 1 1 1 1 y(t) = − e −2t + e− t cos(2t) + e −2t sin(2t). A and B must be complex conjugate of each other and their sum must be real. t ≥ 0 4 4 2 Method : Let Y (s) = Y1 (s) 2 (s + α) + ω 2 (3. In fact. the sum of a complex number (x + jy) and its conjugate (x − jy) is twice the real part 2x = 2Re(x + jy).16) where Y1 (s) is a rational fraction which contains the remaining terms in Y (s).¸ 1 Y1 (−2 − j2)(s + 2 − j2) j2 ¸ ∙ 1 1 −2 − j2 + 1 (s + 2 − j2) = (s + 6) = −Re j2 −2 − j2 + 2 4 −Re Thus Y (s) = − 1 1 s+6 + 4(s + 2) 4 (s + 2)2 + 22 ∙ (3. (3.

24) (3. 30 .. one may use the method familiar from high-school mathematics where one writes simultaneous algebraic equations in the unknown constants by equating coefficients of like powers of s.4 Find the inverse transform of −Re h 1 jω Y1 (−α − jω)[s + α − jω] (s + α)2 + ω 2 i (3.. s + 22 +.so that Y (s) = Example 3.. (3.25) y(t) = 2te−t + e −t − cos(2t) − sin(2t).20) Y (s) = − s2 + s − 10 (s + 1)2 (s 2 + 22 ) (3.22) The second coefficient at s = −1 B= Now obtain C and D Y1 (s) = − h (s2 + s − 10) (s + 1)2 i Y (s) = −Re 1 ( −4+j2−10) (s−j2) j2 ( j2+1)2 s2 + 22 −s − 2 = 2 +.21) Y (s) is first expanded into partial fractions −(s2 + s − 10) A Cs + D B = + 2 + (s + 1)2 (s2 + 22 ) (s + 1)2 s + 1 s + 22 The coefficients are then evaluated as follows ¯ (s 2 + s − 10) ¯ ¯ A =− =2 s2 + 22 ¯s= −1 ∙ ¸ d −s2 − s + 10 =1 ds s 2 + 22 s=− 1 (3. Finally..23) Y (s) = 2 −s − 2 1 + + (s + 1)2 s + 1 s 2 + 22 t≥0 (3.

27) (3.28) The residues of F (s)est at s = −1 is ¸ ∙ 2 (2s + 3s + 3)est = e −t R−1 = (s + 2)(s + 3) s=− 1 ¸ ∙ 2 (2s + 3s + 3)est R−2 = = −5e− 2t (s + 1)(s + 3) s=−2 ¸ ∙ 2 (2s + 3s + 3)est R−3 = = 6e−3 t (s + 1)(s + 2) s=−3 f(t) = L−1 [F (s)] = e−t − 5e− 2t + 6e−3 t Example 3. Associated with each pole of a function of a complex variable is a particular coefficient in the series expansion of the function around the pole. L−1 [F (s)] = Σall poles [residues of F (s)est ] (3.5 Find the inverse Laplace transform of F (s) = 2s2 + 3s + 3 (s + 1)(s + 2)(s + 3) (3.30) .3 INVERSION INTEGRAL METHOD A method of finding the inverse transforms which is distinct from the partial fraction method is the method of residues. The method of residues states: If F (s) is a rational fraction then.29) (3.26) where the residue at an nth order pole at s = s 1 is given by ∙ n−1 ¸ d 1 Rs1 = (s − s1 ) n F (s)est (n − 1)! ds n− 1 s=s1 Example 3.3.6 Find the inverse Laplace transform of 1 (s + α)4 ∙ ∙ ¸ ¸ 1 1 d 3 st L−1 = e (s + α)4 3! ds3 ¯s=− α t d2 st ¯ = (e )¯ ¯ 3! ds2 ¯ s= −α 2 t d st ¯ e ¯ = 3! ds ¯s=− α ¯ t3 ¯ t3 f(t) = est ¯ = e−αt 3! ¯s= −α 3! F (s) = 31 (3.

5 THE SOLUTION OF ORDINARY DIFFERENTIAL EQUATIONS WITH CONSTANT COEFFICIENTS Example 3. A pole on the imaginary-axis (non-zero) gives an oscillatory response component.33) Transforming (3.32) We have previously observed that y(t) shall have terms like e−t .3. e −3t .34) . . . 3. The contribution of zero towards the solution is in the form of amplitude and phase shift. Consider (s + 1) s(s + 2)(s + 3) The location of the poles gives an idea of the nature of the behaviour of the solution. . e−2t. The system-response component due to a pole in the left-half plane dies out after sometime.7 Consider the differential equation d2 x dx + 2x = 0 +3 dt2 dt x(0) = −2. (s − pn ) Where the factors in the numerator are called zeros of the function and the factors in the denominator are called poles of the function. which depend upon the nature of the poles. . x(0) = 1 ˙ s2 X (s) − sx(0) − dx (0) + 3sX(s) − 3x(0) + 2X (s) = 0 dt s 2 X(s) + 2s − 1 + 3sX (s) + 6 + 2X(s) = 0 32 (3. (s − zn ) (3.31) (s − p1 )(s − p2 ) .4 POLES AND ZEROS A rational algebraic polynomial is one whose numerator and denominator can be factorized as below: (s − z1 )(s − z 2 ) . Consider Y (s) = s2 + 2s + 1 (s + 1)(s + 2)2 (s + 3) (3. A pole in the right-half plane results in an ever increasing response.

(s2 + 3s + 2)X(s) = −(2s + 5) X(s) = or. dt dx s 2X (s) − sx(0) − (0) + 4sX(s) − 4x(o) + 5X (s) = 0 dt s2 X (s) − 2 + 4sX(s) + 5X (s) = 0 X(s) = 2 2 = s2 + 4s + 5 (s + 2) 2 + 1 s2 X(s) + 2s − 4 + 4X(s) = 0 x(t) = 2e−2 t sin(t) (3.8 Find the solution of the following differential equation. d2 x + 4x = 0 dt2 for t > 0.40) (3.43) 33 .41) (3.38) (3. x(0) = −2 and x(0) = 4 ˙ s2 X (s) − sx(0) − (3.37) dx (0) + 4X (s) = 0 dt (3. X(s) = X (s) = −(2s + 5) (s + 1)(s + 2) (3.42) −2s + 4 X(s) = 2 s +4 −2s 4 −2s + 4 = 2 + X(s) = 2 s +4 s + 4 s2 + 4 x(t) = −2 cos(2t) + 2 sin(2t) Example 3.9 d2 x dx + 4 + 5x = 0 .39) (3. f or t > 0 dt2 dt dx (0) = 2 x(0) = 0.36) −(2s + 5) (s2 + 3s + 2) 3 1 − s+2 s+1 L− 1[X(s)] = x(t) = e −2t − 3e −t Example 3.35) t≥0 (3.

a) 3. f or t > 0.PROBLEMS 3. for t > 0 + 6 3 + 13 2 + 12 dt4 dt dt dt d3 x d2 x dx d) + 2 2 + 4 + 4x = δ(t). for t > 0 dt3 dt dt all initial conditions zero for (c) and (d). x(0) = x0 dt dx + a 1 x = a 2 δ(t) . x(0) = a. (ii) f(t) = eat u(t) f or t > 0 3. x(0) = 0.3 Find the solution of each of the following differential equations a) dx + a1 x = a2 + a 3 t .1 Find the inverse 1 a) s(s + a)(s + b) 1 c) s[(s + a)2 + b2 ] 2 e) s2 + 3s + 1 s + 5s + 1 Laplace transform of the following functions by partial fractions b) 2 1 s (s + a) d) 2 s 2+ a 2 s (s + b ) s2 + 3s + 1 f) s(s + 1)[(s + 1)2 + 1] 3. (0) = b dt2 dt d4x d3 x d2 x dx c) + 4x = δ(t) . dt2 dt Where. for t > 0.5 Find the time solution of the following differential equation dx d2 x + 4 + 4x = f(t) .2 Find the inverse Laplace Transform using inversion integral for the following s+1 b) a) s + 2 s(s + b) s(s 2 + 1)2 (s + 2) 1 d) c) 2 s 2+ a 2 s (s + a ) (s + 1)5 3. x(0) = x0 b) dt 3. (i) f(t) = au(t).differential equations a) Z dx + 4x + 4 xdt = at dt Z d2x dx b) − x − 2 xdt = u(t) dt +2 dt2 dt where all initial conditions are zero except x(0) = x0 34 .4 Solve the following equations d2 x dx dx (0) = 1 + 5 + 6x = 0 .6 Find the solution of the following integro. dt2 dt dt 2 dx dx b) + w 2 x = 0 .

1) This theorem defines the linearity of Laplace transformation operation.3) (4. 35 Re{s} > c (say) (4.4) 1 = L[(ebt ) − (e− bt )] 2 1 1 1 1 − = 2 (s − b) 2 (s + b) b = 2 s − b2 4.1: If a and b are constants and f(t) and g(t) are transformable functions.2: If f (t) is transformable with transform F (s) then eat f(t) is also transformable.5) Theorem 4.2) L[3t + 5] = L[3t] + L[5] 5s + 3 5 3 = 2+ = s s s2 Example 4. where a is a real constant. Proof: By definition F (s) = Z ∞ f (t)e−st dt . then L[af (t) + bg(t)] = aL[f (t)] + bL[g(t)] (4.2 COMPLEX TRANSLATION (4. Proof: Z ∞ = aF (s) + bG(s) Example 4.1 [af(t) + bg(t)]e− stdt Z ∞ Z =a f (t)e −st dt + b 0 0 ∞ 0 g(t)e−st dt (4.6) 0 .1 LINEARITY Theorem 4. 4.IV LAPLACE TRANSFORM THEOREMS In this chapter we present important theorems which shall help in obtaining the solution of practical problems. and has a transform F (s − a) and conversely.2 L[sinh(bt)] = L [ebt − e−bt] 2 (4.CHAPTER .

3 a: Real translation Right: If f(t) is a transformable function with transform F (s) and a is a non negative real number then f(t − a) more correctly.4 s s 2 + ω2 s+a (s + a)2 + ω 2 L[u(t)] = From the theorem L[eb t] = Example 4.so that by comparison Z ∞ 0 e at f(t)e− stdt ∞ 0 (4.7) Re{s} > c + a = Z e−(s− a) tf (t)dt = F (s − a) . This theorem states that translation in the real domain corresponds to multiplication by an exponential in the transform domain. f(t −a)u(t −a) is a transformable funtion with transform e−as F (s) and conversely.8) 1 (s + a)2 Theorem 4. Proof: Let g(t) = f (t − a)u(t − a) a > 0 then G(s) = = Z ∞ e− stg(t) dt f (t − a)u(t − a) dt 36 (4.3 REAL TRANSLATION 1 s2 1 s 1 s−b (4.5 L[t] = From the theorem L[te−at ] = 4.9) Z0 ∞ 0 .3 L[cos(wt)] = From the theorem L[e− at cos wt] = Example 4. Example 4.

then their transforms are related by the equation. a > 0 L[u(t)] = 1 s e− as s (4. differentiation in the time domain corresponds to multiplication by s in the s domain. first expression on left above vanishes at the upper limit and we obtain L ∙ ¸ df = −f (0) + sF (s) dt (4.10) f(t) = u(t − a) . L ∙ ¸ d f(t) = sL[f(t)] − f(0) dt (4. 4.11) From theorem L[u(t − a)] = L[f (t)] = e −as s Theorem 4.3b: Real translation Left: If f(t) is a transformable function with transform F (s) and a is real non negative for which f(t + a) = 0 for t < 0 .13) −se− stf (t) dt (4. then f(t + a) is transformable with transform easF (s).12) Hence.14) Since f (t) is transformable. v = t − a G(s) = Z ∞ 0 = e− as = e− as F (s) Example 4.4: If f(t) and its derivative dfdt are both transformable functions.6 Now e−s(v +a) f(v)u(v) dv Z ∞ e−sv f(v)dv 0 (4.Now let. Proof: L Integrating by parts = e− stf (t)| 0 − ∞ ∙ ¸ Z ∞ df(t) df = e−st dt dt dt 0 Z ∞ 0 (4.15) 37 .4 REAL DIFFERENTIATION (t) Theorem 4.

n − 1) and the following relation holds. .16) Theorem 4.Example 4.. ..4 can be extended to higher order derivatives. 2. 1.18) dn f f ( n) (t) = n dt £ ¤ L f (n) (t) = s n L [f(t)] − sn−1 f (0) − .19) 38 .7 Let f (t) = eat for t > 0 df = ae at for t > 0 dt f(0) = 1 ∙ ¸ df = sL(eat ) − 1 dt s = −1 s−a then L (4. if L and g(t) = We obtain ∙ ¸ ¯ df ¯ df − ¯ dt dt ¯t= 0 dg = sL[g(t)] − g(0) dt df(t) dt L d2 f dt2 = sL (4. if f(t) and its first derivative exists. where ¯ df ¯ = s[sF (s)] − sf (0) − ¯ dt ¯t=0 ¯ df ¯ = s 2 F (s) − sf (0) − ¯ dt ¯ t=0 (4.. For example. and if the nth derivative is transformable.17) So that. − f(n − 1)(0) n− 1 X n−k−1 (k) = s n L[f (t)] − s n− 1 f(0) − s f (0) k=1 (4. then kth derivatives are also transformable (k = 0. .

4.23) 0 (t − u)f (u) du 39 .5 that ω L [1 − cos ωt] = L [sin(ωu)] s ω2 = s(s2 + ω2 ) Also. 2 is the transform of the function.4 L and dg = sLg(t) − g(0) dt F (s) = sG(s) 1 G(s) = F (s) (4.5.5: Real integration (Definite) If f (t) is transformable.9 F (s) Let f (t) have the transform F (s).22) 0 it follows from Theorem 4. Example 4. The transforms are related by the equation. ¸ ∙Z t 1 (4.21) s This states that integration in the real domain corresponds to division by s in the transform domain. dg dt = f(t) From Theorem 4. s ¸ Z t ∙Z τ f(u) du dτ g(t) = 0 0 1 s ω2 − 2 = 2 2 s ω +s s(s + ω2 ) Changing the order of integration Z t g(t) = = f(u) 0 t Z ∙Z t u ¸ dτ du (4.20) f(u)du = [F (s)] L s 0 Rt Proof: Let g(t) = 0 f (u) du Note that g(0) = 0.8 Since 1 − cos ωt = ω Z t sin(ωu) du (4. its definite Rt integral 0 f(u)du is transformable.5 REAL INTEGRATION Theorem 4. then by Theorem 4. from L[1 − cos ωt] = Example 4.

such as Z t . if f(t) is transformable. . then by Theorem 4. 1 sn+1 F (s) Frequently. .Further.24) If the result of above two examples are tabulated as transform pairs then the extension of the above results gives the following table. let f (t) have the transform F (s). .25) Rt −∞ Summarizing the above. the function Z F (s) is the transform of s3 t b(t) = = = = = g(τ)dτ 0 ¸ Z t ∙Z τ (τ − u)f(u) du dτ 0 0 Z t Zt f(u) [(τ − u)dτ ]du 0 ¸t ∙u Zt (τ − u)2 f(u) du 2! 0 u Zt 2 (t − u) f(u) du 2! 0 (4. its integral f ( −1) (t) = formable and the transforms are related by the equation.5. it is more desirable to deal with indefinite integrals.26) . ¤ £ L f ( −1) (t) = L ∙Z t f (u) du is trans- ¸ f (− 1) (0) 1 f(u) du = L[f (t)] + s s −∞ 40 (4. Transform Function F (s) 1 s 1 s F (s) F (s) . Rt 0 f(t) Rt f(u) du 0 Rt f(u) du 0 (t−u)n n! f(u) du g(t) = or g(t) = = Z t f(u) du −∞ t f (u) du = −∞ t Z Z f(u) du + 0 Z 0 f(u) du −∞ f (u) du + g(0) 0 g(0) 1 L[g(t)] = L[f (t)] + s s (4. .

4.29) and Example 4. tf(t) is also transd formable and has the transform − ds [F (s)].6: If f(t) is transformable with transform F (s).28) 1 s ∙ ¸ 1 d 1 = 2 ds s s ∙ ¸ d 1 2! L[t2 ] = − = 3 ds s2 s L[t] = − L[tn ] = (−1)n ∙ ¸ n! dn 1 = n+1 dsn s s ω s2 + ω2 (4.2 L[u(t)] = It follows that dF ds (4.30) . F (s) = Z ∞ 0 e−st f (t) dt (4.11 L(sin wt) = L(t sin wt) = − ¸ ∙ d ω ds s2 + ω2 2ωs = 2 (s + ω2 )2 41 (4.10 Using the result of Example 4.27) or Z d ∞ −st dF = e f(t) dt ds ds 0 Z ∞ d −st (e )f (t) dt = ds 0 Z ∞ = −te− stf (t) dt Z0 ∞ = e− st(−tf (t)dt) 0 L[tf (t)] = − Example 4. Proof: By Definition.6 COMPLEX DIFFERENTIATION Theorem 4.

33) (4. then the transform of f (tt) is related to it by the equation. the transform will also (in general) be a function of that parameter.12 ∙ ¸ 1 d ds s + α 1 = (s + α)2 L[te− αt] = − (4. α) is transformable.8 SECOND INDEPENDENT VARIABLE Suppose for a particular α.7: If both f(t) and f (t) are transformable functions and the transform t of f (t) is F (s). 42 .Example 4. Proof: By definition F (s) = Z ∞ 0 F (ξ)dξ = Z Z s ∞ 0 e−st f (t) dt ∞ ∞ ∙Z 0 ¸ e −ξt f(t) dt dξ Assuming the validity of changing the order of integration Z ∞ ∙Z ∞ ¸ = f(t) e−ξt dξ dt ∙ s ¸∞ Z0 ∞ e −ξt f(t) dt = −t s ∙ ¸ Z0 ∞ f (t) e −st dt = L f(t) = t t 0 Example 4. f(t.7 COMPLEX INTEGRATION Theorem 4.13 L(sin ωt) = It follows that L ∙ ¸ ¯∞ Z ∞ sin ωt ω ξ¯ = dξ = tan−1 ¯ t ξ 2 + ω2 ω ¯s s π ω −1 s −1 s = cot = tan− 1 = − tan 2 ω ω s ω s2 + ω 2 (4. Z ∞ 1 F (ξ)dξ (4.34) 4.32) L f (t) = t s Hence division by t in the real domain is equivalent to integration in the transform domain.31) 4.

35) ∂ ∂ F (s.That is. = = − ∂α s + α (s + α)2 43 (4.14 Since. α) = If α is made to vary. α)] = L f(t. α)dα α0 α0 (4. α) ∂α ∂α ∙Z α1 ¸ Z α1 L [f(t. α) is transformable function with respect to t with α a second independent variable. Z ∞ lim F (s.37) Example 4. α) = lim e−st f(t. L[e−αt ] = it follows from Theorem 4. α) dt α→α 0 α→α 0 0 Z ∞ = e −st lim f (t.8 that L[u(t)] = L [limα→0 e−αt ] 1 1 = = lim α→0 s + α s L[te −αt ] = L ∙ 1 s+ α (refer to Example 1.38) ¸ −∂ −αt e ∂α 1 1 ∂ .36) Theorem 4. α) dα dt α0 (4. α) dt ∂α α F (s. α)] dα = L f(t. α) dt 0 α→α 0 Z ∞ 0 e−st f (t. α) dt (4. ∙ ¸ lim L[f(t. F (s.39) . then under ‘suitable’ conditions.2) (4. α)dα = α0 ∞ = Hence the following theorem Z α0 0 ∞ − st e 0 e −st f(t. α) dt dα ¸ ∙Z α f(t. α)] = L lim f (t. then under suitable conditions the following relation hold. α) = ∂α ∂α and Z α Z ∞ 0 e− stf (t. α) α 0→α α→α 0 ∙ ¸ ∂ ∂ [Lf(t. α) dt = Z Z Z ∞ 0 e−st ∂f(t.8: If f (t.

. (k−1 )T (k+ 1)T e− stf (t)dt kT Making the change of variable t = u + kT .45) .9: If f(t) is transformable periodic function of period T . + 0 Z kT e−st f (t)dt + .42) (4.9 PERIODIC FUNCTIONS A function f(t) is periodic of period T if f(t + T ) = f(t) for every t.41) Theorem 4.L ∙ 1 − e− αt t ¸ =L ∙Z α e− βt dβ = − Z 0α 0 1 dβ s+β α ¸ ∙ s+α s ¸ = log(s + β)| 0 = log (4.43) 1 − e−st Proof: Z ∞ 0 e− stf (t)dt = Z T e−st f(t)dt + 0 k=∞ = XZ k=0 Z 2T e −st f(t)dt + . R T −st e f(t) dt L[f(t)] = 0 (4.44) Z Z e− su f(u) du 0 T e−skT ! 0 e−su f (u) du 1 − e −st 44 (4. then its transform may be found by integration over the first period according to the formula. . f (t + 2T ) = f(t + T + T ) = f(t + T ) = f (t) it follows that if f(t) has period T . . f(t + kT ) = f(t) (4. Since. .40) 4. so that Z ∞ 0 dt = du ∞ XZ k=0 T e−st f(t)dt = = = −0T e−s( u+kt) f(u + kt) du à ∞ X k=0 (4.

Example 4.46) 45 . . . L[f (t)] = = = = = f(t)e− st dt −2s 0 1 − e Z 1 e−st dt 1 − e−2s 0 ¯1 − 1 e− st ¯ s ¯ 1 − e−2s ¯0 (1 − e −s ) s(1 − e−s )(1 + e− s) 1 s(1 + e−s ) Z 2 (4.1 Let f (t) be the square wave n 1 for 2n ≤ t < 2n + 1 f (t) = 0 for 2n + 1 ≤ t < 2n + 2 n = 0. 2.15 Consider the periodic function of Fig 4. . 1.

47) L f( ) = aF (as) a Hence the division of the variable by a constant in the real domain results in multiplication of both the transform F (s) and the transform variable s by the same constant. then ¸ ∙ t (4. we obtain the first half period by shifting sint to the right by π units and by adding the result to sint. f (t) = u(t) − 2u(t − 1) + u(t − 2) L[f (t)] = 1 2e −s e− 2s − + s s s −s 2 (1 − e ) = s k= 0 Define the square Theorem 4.2 and transform it into Laplace domain. so that f1 (t) = sint + sin(t − π)u(t − π) Lf1 (t) = F (s) = and L |sin t| = 4.10 CHANGE OF SCALE Theorem 4.Example 4. 4.17 The function f(t) = |sint| is a rectified sine wave and is periodic with period T = π. We make the change in the variable τ= in the integral definition of the transform F (s) = Z ∞ 0 Then 1 + e−π s s2 + 1 1 + e−π s (s + 1)(1 − e−π s ) 2 t a f (τ)e −τ s dτ (4. A single period of f(t) corresponds to a half period of the function sint.9 wave as f1 (t) = P∞ f (t − 2k) with a period of T = 2.48) 46 .10: If the function f (t) is transformable and L[f (t)] = F (s) and a is positive constant (or variable) independent of t and s. so that by (1 − e −s )2 s(1 − e −2s )2 L[f1 (t)] = F1 (s) = Example 4.16 Consider the waveform of Fig. Using the symmetry of sint.

05)2 + 10−2 4.11 REAL CONVOLUTION Suppose f (t) and g(t) have transforms F (s) and G(s) respectively. at u = 0 t = α. Is there a function h(t) with transform F (s). This is accomplished by letting a = 103 . we obtain F (as) = Rearranging Z ∞ 0 t −ast t f( )e a d( ) a a ∞ (4. Proof: Z ∞ e−sv f(v) dv e−su g(u) du 0 0 Z ∞Z ∞ = e−s(u+v) f (v)g(u)dudv ∞ 0 0 F (s)G(s) = Z (4.05 = (4.After changing the variable. Suppose we wish to find the transform pairs in which t is measured in milliseconds. the product F (s) x G(s) is the transform of Z t h(t) = f(v)g(t − v) dv (4.54) For a fixed v let t = u+v dt = du t = v.18 Given the transform pair Z 0 t f ( )e− st dt a (4.49) aF (as) = Example 4.50) s + 50 = L [e− 50t cos 100t] (s + 50)2 + 104 in which t is in seconds. G(s) and if so.52) 0 h(t) is termed the real convolution of the functions f(t) and g(t) and is usually written as f(t) ∗ g(t) h(t) = f (t) ∗ g(t) = g(t) ∗ f (t) (4.53) Since the convolution is symmetric. how is it related to f (t) and g(t)? Theorem 4.51) (s + . at u = ∞ 47 . ¸ ∙ h −50t i 1000s + 50 L e 1000 cos(10− 1 t) = 1000 (1000s + 50)2 + 104 s + 0. This theorem states that the product of two functions of s is the Laplace transform of the convolution of the two respective functions of time t.11: If f(t) and g(t) have transforms F (s) and G(s) respectively.

19 L[t] = and 1 s2 has the transform ω L[sin ωt] = 2 s + ω2 Z t h(t) = (t − u) sin ωudu Z0 t u sin ω(t − u)du = 0 w s 2 (s2 + ω2 ) ¯t Z t u cos ω(t − u) ¯ cos ω(t − u)¯ − du h(t) = ω ω 0 0 ¯t t sin ω(t − u) ¯ ¯ = t − sin ωt = + ¯ ω ω2 ω ω2 0 1 1 H (s) = 2 − ωs ω(s 2 + ω2 ) ω H(s) = 2 2 s (s + ω2 ) H(s) = (4.56) (4. 2 = L[cos t] s+ a s +1 48 . since g(t − v) = 0 for t < v ¸ ∙Z ∞ Z ∞ = e− st f (v)g(t − v) dv dt 0 Z0 ∞ − st e h(t)dt = 0 0 0 Z 0 ∞ Z0 ∞ Z ∞ Z ∞ e−st f (v)g(t − v)dtdv where h(t) = Z ∞ 0 f(v)g(t − v)dv = Z t 0 f (v)g(t − v)dv.20 We are given the transform F (s) = s (s + a)(s2 + 1) and are to find the inverse transform f(t).55) Example 4.57) Example 4. From the known pairs s 1 = L[e− at] . since g(t) = 0 for t < 0 (4.so that F (s)G(s) = = e−st f (v)g(t − v)dvdt.

59) 4.12: If f (t) and its derivatives are transformable.13: This theorem states that the behaviour of sF (s) near the point at infinity in the s-plane corresponds to the behaviour of f (t) near t = 0. and if f (t) has the transforms F(s) and if all the singularities of F (s) are in the left half plane.12 FINAL VALUE THEOREM Theorem 4. (4. then lim sF (s) = lim f(t) s→0 t→∞ This theorem states that the behaviour of F (s) near the origin of the s-plane corresponds to the behaviour of f(t) for large t(t → ∞). (See Eq.21 Let F (s) = 1 s(s + α) 49 .60) lim sF (s) = f(0) (4.We deduce that f(t) = e− at ∗ cos t = e−a(t−u ) cos u du Z t eau cos u du = e−at 0 0 Z t = 1 [a cos t + sin t − ae−at ] a2 + 1 (4.13)) dt lim sF (s) = lim f(t) s→∞ t→∞ (4.13 INITIAL VALUE THEOREM Theorem 4. Proof: df e−st dt + f(0) dt Z∞ df lim sF (s) = lim e −st dt + f(0) s→∞ s→∞ 0 dt Z ∞ df −st = dt + f (0) lim e s→∞ dt 0 sF (s) = 0 s→∞ Z ∞ (4.58) 4.61) Example 4. Proof: lim F (s) = s→0 df + f(0) dt ∞ = f(t)| 0 + f(0) = lim f (t) − f(0) + f (0) = 0 t→∞ Z ∞ 0 ∞ lim e−st s→0 Z df dt + f(0) .

4.then its real and imaginary parts are transformable and the operation of transforming and taking real and imaginary parts is commutative.22 ω s2 + ω2 ωs sF (s) = 2 s + ω2 F (s) = This does not satisfy the condition of Theorem 4.14 EXTENSION TO COMPLEX FUNCTIONS Theorem 4. L[Ref(t)] = ReL[f(t)] L[Imf(t)] = I mL[f(t)] Example 4. It follows from linearity of functions that.14 L[e jωt ] = L[cos ωt + j sin ωt] = L[cos ωt] + j(L[sin ωt]) since s s 2 + ω2 ω L[sin ωt] = 2 s + ω2 L[cos ωt] = L[e jωt] = s jω + s 2 + ω2 s 2 + ω2 s + jω 1 = 2 .63) Therefore.14: If f (t) is a complex valued transformable function.23: Consider the following transform pair L[ejωt ] = But by theorem 4.sF (s) = lim sF (s) = s→0 1 s+α (4. (4. ω real s − jω s + ω2 (4.62) 1 = lim f(t) α t→∞ Example 4.64) 50 .12 since its singularities lie on the imaginary axis.

8 Use Theorem 4.6 Starting with the transform pair ∙ µ ¸ ¶ 1 e− αt 1 −β L− 1 + √ 2 sin βt − tan− 1 = 2 s[(s + α)2 + β 2 ] α + β2 β α + β2 α Use Theorem 4.9 to find the Laplace transforms of the function shown in Fig.4 Since L[cos ωt] = s s2+ω 2 .6 to obtain L [te−at sin ωt] from L[sin ωt] = 4.2 From the knowledge that L[3t + 5] = 5s+3 s2 .5 Use Theorem 4.PROBLEMS 4. use Theorem 4. (t − a)u(t − a) 4.2 to find L [(3t + 5)e−t ] 4.1 Use Theorem 4.1 to find the Laplace transform of the functions a + b + ct2 4.3 Find the Laplace transform of the ramp function translated a units (a > 0) to the right.8 to derive the following a) L− 1 [ 1 ] s h i 1 −1 b) L ( s+α)2 c) L− 1 [ s1n ] 4.3 51 .2 and 4.8 to derive the following ∙ L− 1 4. use Theorem 4. 4.6 to derive L[t cos ωt] ω s2+ω 2 4.7 Starting with the transform pair L−1 1 = e−αt s+α ¸ 1 s(s + α)2 Use Theorem 4.

a1 .11 If L[f(t)] = F (s) find the transform of the following functions a) f (t)u(t − α) b) f (t)[u(t − α) − u(t − β)] Assume that α and β are positive real numbers 4. = u(τ )e−α(t− τ ) dτ s s+α 0 ¸ Z t ∙ 1 1 .9 Use convolution integral to find the following transform a) L−1 ¸ Z t 1 1 .10 Assuming all poles lie to the left of the imaginary axis.4. a3 . Without carrying out the inverse transform find: a) The form of each term b) The initial value of e0 c) The initial values of the first two derivatives of e0 d) The final value of e 0 52 . = e−jβ τ e +jβ(t− τ ) dτ b) L− 1 s + jβ s − jβ 0 ∙ 4.12 Find the inverse Laplace transform of the following functions e −s 2 a) (1 − s ) b) 1 − e−s s2 (1 + e −s ) 4. All constants(a0 . find the final and initial values of the time function whose transform is F (s) = k(s + d) s 3 + a2 s2 + a1 s + a0 4. β) are positive and real.13 The Laplace transform of the output voltage of an amplifier is given by L[e0 (t)] = a3 s 3 + a2 s2 + a 1 s + a0 s2 (s + α)[(s + α)2 + β 2 ] The input is a unit impulse. α. a2 .

2) (5.5) .V S-PLANE ANALYSIS 5.4) (5. The method automatically includes the initial conditions.CHAPTER . Assuming x(0) = 0. x(0) = 0.3) giving r B K and ξ = √ M 2 KM ωn = The two parameters ξ and ω are sufficient to describe the second order equation. many problems can be solved without the labor of inverting the Laplacetransformed equations. (5. In the s-plane analysis. 5. the Laplace transformed equation is ˙ (s2 + 2ξωn s + ω2 )X(s) = F1 (s) n The ratio X (s)/F (s).1 INTRODUCTION The Laplace transform offers a method of finding the solution of linear differential equations with constant coefficients. is X(s) 1/M = 2 F (s) s + 2ξωn s + ω2 n 53 (5. defined as the transfer function of the system.2 SOLUTION OF A SECOND ORDER SYSTEM M d2 x dx + K x = f(t) +B dt2 dt Let. ξ = damping ratio ωn = undamped natural f requency The linear differential equation can be written as d2 x f(t) dx 2 + ωn x = f1 (t) = + 2ξωn dt2 dt M Comparing the coefficients of Eq.1) (5. (5.1) and Eq. 2ξωn = B M K ω2 = n M (5. The necessary design information can be obtained by locating the roots of the characteristic equation on the s-plane and by observing how these roots vary as some parameter is changed.2).

the RHS of Eq.For F (s) = 1 (f(t) = δ(t)).1. 5. √ 2 1 ( −ξω n+jω n√ 1−ξ 2)t 1 e + e(−ξ ωn −jωn 1−ξ ) t M M p 2 − ξω nt e = cos( 1 − ξ2 ωn t).7) (5. 5. the characteristic equation can be written s2 + and the roots are si = − B ± 2M Bs K + =0 M M µ B2 K − 4M 2 M ¶1 2 (5.8) equal to zero and is found from the expression .2 (a). (5. ξ = 0.95 54 . 5. t ≥ 0 M x(t) = (5.1 (undamped) Roots of the characteristic equation at s = −1 ± j9.6) The impulse response is plotted in Figs. Critical damping in the second order system is defined as the value of damping which produces two equal roots in the characteristics equation and separates sub-critical and super critical regions. and the quantity ξ is the ratio of damping that exists in the second order system to critical damping. (5. Fig.1 (a)-(c) for ωn = 10. 2 The characteristic equation is s2 + 2ξωn s + ωn = 0.1 (a) Impulse Response of the vibration table undamped natural frequency ωn = 10. 1. damping ratio ξ = 0.6) gives the impulse response of the system. For the system of Fig.8) Critical damping occurs for the value of B = Bc which makes the term under the radical of Eq. 2.

Fig..32 55 . −37.68. roots at s = −2. Double roots at s = −10 Fig. 5.1 (b) Impulse Response of the vibration table ωn = 10. ξ = 2 (over damped).1 (c) Impulse Response of the vibration table ωn = 10. 5. ξ = 1 (critically damped).

.

11) Bc /2M 2 KM The undamped natural frequency is the frequency of oscillation that occurs with zero damping. In this case. and the damping ratio is cosθ = ξ.12) si = ± M M q K Then the undamped natural frequency ωn = M The roots of the second order equation are located on the s-plane as shown in Fig.9) √ Bc = 2 K M (5. At ξ = 0. In higher order systems. these roots move along a semicircle. 5. it is not necessary to continue unless we wish to determine the actual number of roots in the right half plane.3 ROUTH HURWITZ STABILITY CRITERION This method provides a simple and direct means for determining the number of roots of characteristic equation with positive real parts (i. For ξ > 1 (overdamped). both roots are real.2 (b). If any power is missing or any coefficient is negative.13) For a stable system. The Routh-Hurwitz method centers about an array which is formed as follows: 57 . + a1 s + a0 (5. Although we cannot actually locate the roots. roots which lie in the right half s-plane).10) Bc For this value of B. if B = 0. From the definition. (5.e. there exists a double real root at − 2M . 5. we find the damping ratio to be B B/2M = √ ξ= (5. as ξ is varied from 0 to 1. For constant ωn . the roots are located on the j ω axis and when ξ = 1 the roots are on the real axis. The transfer function of a linear system is a ratio of polynomials in s and can be written: H(s) = N(s) N (s) = n D(s) s + an− 1 sn− 1 + an−2 sn−2 + . we can determine without factorizing the characteristic equation.2 Bc K =0 − 4M 2 M (5. are r r −K K = ±j (5. The radius is the natural frequency ωn . where there are more than two roots.7). and the system is unstable or marginally stable (necessary condition). . we know immediately that D(s) has roots in the right half plane or on the jω − axis. if any of the roots lie in the right half plane and hence give rise to an unstable system. all the a0 s are positive real constants and all powers of s in the denominator polynomial are present. the system response is often dominated by two ‘least damped’ roots and the results of the second order system can be extended to approximate the higher order systems. The roots of Eq. .

15) We continue the pattern until the remaining b0 s are equal to zero.. . . . from the Routh Hurwitz Criterion: The number of roots of the characteristic equation with positive real parts is equal to the number of changes of sign of the coefficients in the first column. .p. . . .. . an−7 . .. (5. The remainder of rows down to s is formed in a similar fashion. . .h. .. Example 5. . . . . .. if all the terms of the column have the same sign. b4 .. . . Hence. . . . .1 Consider the following polynomial D1 (s) = s5 + s 4 + 3s3 + 9s2 + 16s + 10 58 (5.... .16) This is continued until all remaining c0i s are zero. . These constants are evaluated as follows: c1 = b1a n− 3 − b2 an− 1 b1 ba − b3 an− 1 c2 = 1 n− 5 b1 b1a n− 7 − b4 an− 1 c3 = b1 c4 = ... Having formed the array. Each of the last two rows contain only one non-zero term. . . .. the system is stable. b1 = (5.. ...17) . . we can determine the number of roots in the r. . . . . an−6 . .. . an− 4 an− 5 b3 c3 . . . j1 . sn−2 sn−3 s 1 s 0 where the constants in the third row are formed by cross multiplying as follows: an−1 an−2 − an an−3 an− 1 an−1 an−2 − an an−5 b2 = an− 1 an−1 an−6 − an an−7 b3 = an− 1 b4 = . . using the sn− 1 and sn−2 rows..sn s n−1 an a n−1 b1 c1 i1 . The next row is formed by cross multiplying. . an−2 an−3 b2 c2 .

2. Example 5.18) 59 . Special Case 1: When the first term in a row is zero. In the original polynomial. Replace the zero with a small positive number and proceed to compute the remaining terms in the array.We form the Routh Hurwitz array s5 s 4 1 1 (1)(3)− (1)(9 ) 1 3 9 = −6 = 10 = 12 (1) (16)− (1)(1 0) 1 (−6 )(10 )−0 −6 16 10 = +6 s3 s2 s 1 (− 6)(+9)−(1 )(6) −6 (10)(6) −(10 )(−6 ) 10 = 10 s 0 10 There are two changes of sign in the first column from +1 to -6 and from -6 to +10. Two special cases may occur: 1. This is also the number of roots of s with positive real parts. Therefore. we conclude that there are two roots in the right half plane. but other terms in the row are not zero. 2. the coefficients of any row may be multiplied or divided by a positive number without changing the sign of the first column. one of the following two methods can be used to obviate the difficulty. In order to avoid labor in the calculations. substitute 1 for s and find the number of roots of x which have x positive real parts. A zero in the first column. 1. A row in which all coefficients are zero.2 We illustrate the above methods on the polynomial D(s) = s 5 + 3s4 + 4s3 + 12s 2 + 35s + 25 Method 1: The array is as follows (5.

s5 s 4 1 3 0 4 12 80 3 80 3 35 25 s3 s3 s2 s s 1 12 − 8 0 [(12 − 25 80 8 0 3 25 ) − 25 ]/[12 − 80 ] 0 The first term in the s 2 row is negative as approximately s5 + s4 s 3 + 80 3 → 0 and the first term in the s1 row becomes . 5 x and form the array x5 x x 4 (5. there are two roots in the right half plane. Method 2: We replace s by 1/x in the original polynomial 1 1 1 1 1 1 D( ) = ( )5 + 3( )4 + 4( ) 3 + 12( ) 2 + 35( ) + 25 x x x x x x 1 = (25x5 + 35x4 + 12x3 + 4x2 + 3x + 1) . The signs of the first column are + + − + + s2 s1 s 0 There are two changes of sign.19) 25 35 320 35 12 4 80 35 3 1 3 x3 x2 x1 x0 4 − 19 4 35 19 1 1 1 60 . hence.

replaced s0 4 The existence of all zero s1 row indicates the presence of equal roots of opposite sign..20) 1 4 11 2 11 22 24 80 3 2 20 11 30 24 8 8 3 0 50 11 s3 multiply by 11 50 1 s 2 4 8 4 divide by 2 2 1 s 1 0 {2} . The all zero row is replaced by the coefficients obtained by differentiating an auxiliary equation which is formed from the previous row. The roots of the auxiliary equation which are also the roots of the original equation occur in pairs and are the opposite of each other. any time the polynomial has two equal roots with opposite signs or two pairs of complex conjugate roots.Since there are two changes of sign.21) (5. Example 5. we make use of the following procedure: 1. Special Case 2: When all coefficients in any one row are zero. . however. The occurrence of all zero row usually means that two roots lie on the jω axis. there are two roots of x with positive real parts. This condition occurs. The coefficient array is formed in the usual manner until all zero coefficient row appears.22) . 2. (5. there are two roots of s with positive real parts. .. Hence. The auxiliary equation is s2 + 4 = 0 Differentiating Eq.21) 2s + 0 = 0 61 (5.3 Consider the following polynomial: F (s) = s6 + 4s5 + 11s 4 + 22s 3 + 30s2 + 24s + 8 The coefficient array is written s6 s5 s s 4 (5. .

4 As a practical example of the use of the Routh Hurwitz method. 5. we need not continue further. there is no root which has positive real part.3.24) R(s) τaτm s 3 + (τa + τm )s 2 + s + aKsK m where A = Ks = Km = τm = τa = C(s) = R(s) = The amplifier gain Potentiometer sensitivity Volts/radian Motor constant. The roots are s = ±2j (5.25) Since Eq. (5. sec Laplace transformed output position Laplace transformed input position Suppose we wish to determine the effect of the amplifier time constant τa upon the system stability.23) These roots give rise to an undamped sinusoidal response. The Routh Hurwitz array is established s3 s s 2 τa τm (τa + τm ) (τa + τm ) − AKs Km τaτm τa + τ m 1 AKs Km 1 We know that if all the coefficients in a row are zero. consider the servomechanism of Fig. whose block diagram is given in Fig. We can obtain all zeros in a single row by setting the first term in the s1 row equal to zero with the result τa τ m 1 = KK A τa + τm s m (5.e. 5.25) gives us the relation between A and τa .So the coefficient pf 0 s0 is 2 Since there are no changes of sign in the first column.5 which shows the effect of τ upon system stability. The overall transfer function of the system is C(s) AKs Km = (5. two equal imaginary roots). Example 5. radians/Volt/Sec Motor time constant Amplifier time constant. 5. Eq. 62 . (5. we have two equal roots of opposite sign. Note that the larger the amplifier time constant the smaller the amplifier gain (for stability).4. To do this we shall find the relationship between the variables for marginal stability (i.25) is plotted in Fig.

.

where K is the constant portion of the loop gain.29) (5. (5.34) is used to determine the gain k at points along the locus.28) Hence the entire KGH function is a complex quantity and is written in polar form as K(A1 ejφ 1 )(A3 e jφ3 ) (A e )(A2 ejφ 2)(A4 ejφ 4 ) K A1 A3 j[(φ1+φ 3)−( nφ 0+φ2+ φ4)] = n e A0 A − 2A4 jφ = Ae n jnφ0 0 KGH = (5. G(s) is the forward loop transfer function.30) (5.33) (5.6).27) Each term in KGH is a complex number and may be written in polar form as s+ 1 = A1 e jφ1 τ1 (5. 3 .33) and (5. KGH = which may be rewritten in the form K1 τ1 τ3 (s + τ11 )(s + τ13 ) τ2 τ4 s n (s + τ12 )(s + τ14 ) K1 (sτ1 + 1)(sτ3 + 1) s n (sτ2 + 1)(sτ4 + 1) (5. . The root locus traces the location of the poles of the closed-loop transfer function C (s)/R(s) (zeros of 1 + K G(s)H(s) = 0) in the s-plane. (5. 2. these are easily determined from the loop transfer function which for the root locus method is written as K GH = (G(s)H(s)). and H(s) is the feedback loop transfer function. The loop transfer function KGH is written as a ratio of factored polynomials e..26) K GH = (5. 5. and Magnitude of KGH = |KGH| = A = 1 (5. With the locus plotted Eq.4 ROOT LOCUS ANALYSIS The root locus method is based upon the knowledge of the location of the roots of the system (Fig.g. 1. as K is varied from 0 to ∞.33). The algebraic equation from which the roots are determined is 1 + K GH = 1 + Aejφ = 0 (5. (5.34) are the result of setting 1 + KGH = 0. .32) which furnishes the two expressions Angle of K GH = arg KGH = φ = (2k + 1) 180 k = 0. The locus is plotted by finding all points s which satisfy Eq. 64 . In most cases. with the feed back loop opened H(s) = 0.31) Where K = K− 1φ1 φ3 φ 2φ4 .34) Eqns.5.

. R(s) = C(s) = B(s) = E(s) = A(s) = G(s) = Refrence input Output (controlled variable) feedback signal R(s) − C(s)= error signal actuating signal C( s) = open-loop transfer function A(s) C(s) G(s) G(s) = = = closed-loop transfer function R(s) 1 + G(s)H(s) 1 + K G(s)H(s) H(s) = feedback path transfer function KG(s)H(s) = loop transfer function 65 .

· · · where p is the number of poles and z is the number of zeros.8 illustrates Rule 3 for the above example. and s = −2 ± j 15. for K = 0. Rule 3. 5. Continuous curves which comprise the locus start at each pole of K GH. there exist four branches. 1. 1. and the asymptotic angles are computed as follows: θk = (2k + 1)180 deg k = 0.) Rule 1. KGH = K(s + 10) s(s + 5)(s + 2 + j15)(s + 2 − j15) (5. K = 2 Rule 4. 5. The rule can be expanded to read. Any complex poles and zeros are ignored while applying this rule. Here p = 4.35).35) (The poles and zeros of Eq. three of the branches must terminate at infinity. K = 1. 2 p−z θ1 = 180 deg θ2 = −60 deg (5. In Fig. The locus exists at any point along the real axis where an odd number of poles plus zeros K GH are found to the right of the point. terminate on the zeros of KGH for K = ∞. For large values of gain.35) are shown in Fig.The rules that are stated and demonstrated in this section enable the engineer to sketch the locus diagram rapidly.7 the locus exists along the real axis from the origin to the point at s=-5 and s=-10 to minus infinity (see Fig. The branches of the locus which are single valued functions of gain. z = 1 66 . (5.11) Since there is only one finite zero at s = −10. Fig 5. In Eq. the branches of the locus are asymptotic to the angles (2k + 1)180 deg p−z k = 0. (see Fig. If the number of poles exceeds the number of zeros. (5. The gain K is usually positive and varies from 0 to ∞ . The following loop transfer function is used to demonstrate the method. 5. For the example Σpoles = (−5) + (−2 + j15) + (−2 − j15) + 0 = −9 Σzeros = −10 p = 4. starting from the poles located at s = 0. z = 1.11). that locus starts at poles and terminates on either a finite zero or zeros located at s = ∞. s = −5. some of the branches will terminate on zeros that are located at infinity. The starting point for the asymptotic lines is given by CG = Σpoles − Σzero p −z which is termed as the centre of gravity of the roots.36) θ0 = 60 deg K = 0. Rule 2. 5. The angular directions to which the loci are asymptotic are given by Rule 3. 2.7.

31) is found from the equation: 1 1 + n σb σb + + 1 σb + += 1 σb + + 1 σb + τ13 (5. (5.7 is redrawn in Fig. Rule 5. The angle of departure from the complex poles and the angle of arrival to complex zeros is found by using Eq. If n root loci are involved at a breakaway point . The gain K at a point s1 is evaluated from the criterion of Eq. the system becomes unstable at this value of K.38) K= 1 |G(s 1 )H(s1 )| 67 (5.7 deg +θ5 ) + 62 deg = 180 deg The root locus procedure is based upon the location of the poles and zeros of KGH in the s-plane.37) The asymptotic lines found in Eq.9 Since the complex poles and zeros always appear in conjugate pairs i. 5. −14. σb = −2. The pole-zero configuration of Fig. equal vertical distances from the real axis. In the example. −(θ 1 + θ2 + θ 3 + θ5 ) + θ 4 = 180 deg −(97. (5.. the centre of gravity always lies on the real axis.36) start from the centre of gravity. If the locus crosses the imaginary axis for some gain K. These lines are placed on the s-plane as shown in Fig.05 Rule 6.33). They are merely the terminal points for the loci of the roots of 1 + K GH.CG = (−9) − (−10) 1 = 3 3 (5. 5. The degree of stability is determined largely by the roots near the imaginary axis.39) or .35) is shown in Fig. 5. (5. Two roots leave (or strike) the axis at the break away point at an angle of ±90 deg. The root locus sketch for the system of Eq. the values of gain that occur at certain points along the locus must be found. (5. The break away point for real axis roots Jb for Eq.954.10 for the purpose of finding the angle of departure from the complex pole at s = −2 + j15. (5. These points do not move.11. Rule 7.e. The rules for sketching the locus are based on the angle criterion of Eq. they be 180/n degrees apart.38) 1 τ2 1 τ4 1 τ1 where σb + 1/τ1 is the magnitude of the distance between the assumed break away point σb and the zero at −1/τ1 .6 deg +90 deg +78. 5.33) which is repeated here argK GH = (2k + 1)180 deg After the locus is sketched and certain points located more accurately. The angles subtended by the poles and zeros to the pole in question are added (positive for zeros and negative for pole). (5. The initial angle of departure of the roots from complex poles is helpful in sketching root locus diagrams.

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where |GH | is the product of the magnitudes of the distances from the zeros to the point at which the gain is to be evaluated. K ≈ 194 at the breakaway point s = −2. If no zeros are present in the transfer function. the product of the zero distances is taken equal to unity.95.40) The magnitudes are measured directly from the root locus plot. divided by the product of the distances to the poles. Thus K= product of pole distances product of zero distances (5.. In the example. At the point where the locus crosses the imaginary axis. This value of K (on the verge of instability) can also be determined from the Routh Hurwitz criterion 69 . K = 751.

ii) a < 0. determine the values of K which correspond to a stable system. By means of Routh Hurwitz stability criterion. the left half plane. Find Y (s). 5.4.6. y(0) = 1.PROBLEMS ˙ 5.6) the open-loop transfer function G(s) = K s(s + 1)(s + 3. and on the jω axis. 5.6.1. Discuss stability. 5.2. 5. for f(t) = 0. Sketch the root-locus diagrams of the system in Fig. Discuss stability. 5.5. The characteristic equations for certain systems are given below.H = s 2 + 2s + 100 s s+4 K (s + 2) . The quantities KG and H are defined below. Consider the system equation y(t) + a(y(t) = f(t). For each of the following transfer functions locate the zeros and poles.3. Draw the root locus of a unity-feedback system with (H(s) = 1in Fig.H = 2 b) K G = s(s + 20) s 70 . obtain y(t) by inverse Laplace transforming Y (s). determine the stability of the systems which have the following characteristic equations: a) s 3 + 20s 2 + 9s + 100 = 0 b) s4 + 2s3 + 6s 2 + 8s + 8 = 0 c) s6 + 2s 5 + 8s4 + 12s3 + 20s2 + 16s + 16 = 0 In each case.5)s2 + 4K s + 50 = 0 In each case. Draw root locus G(s) Discuss the stability of each case. determine the number of roots in the right half plane. a) KG = K 1 . Locate the poles of Y (s) in the s-plane. a) s 4 + 22s 3 + 2s + K = 0 b) s3 + (K + 0. Plot y(t) for i) a > 0. sketches for the closed-loop system 1 + G(s) K s(2s + 1) K(s + 1) b) G(s) = 2 (s + s + 10) K c) G(s) = s(s + 1)(s2 + s + 10) a) G(s) = 5.5)(s + 3 + j 2)(s + 3 − j 2) 5.

electric circuits and mechanical vibrations. there results 1 f(x) cos nx = a0 cos nx + . such as. engineers and mathematicians from both an applied and theoretical point of view. It is convenient to assume that the series is uniformly convergent. heat conduction.CHAPTER . when m 6= ±n 71 (6.1 EULER-FOURIER FORMULAS A function f(x) can be represented by a trignometric series as follows: X 1 f(x) = a 0 + (an cos nx + bn sin nx) 2 (6.2) f(x)dx = a0 π (6. Since that time Fourier series and generalization to Fourier integrals and orthogonal series. −π (6. if we multiply Eq.J. This trigonometric series is now required in the treatment of many physical problems. in general −π (6. Since Z the calculation yields Z π π cos nxdx = −π Z π sin nxdx = 0 for n = 1. electromagnetic waves. π) and coefficients an and bn are to be found.1)by cos nx.VI FOURIER SERIES INTRODUCTION In the early years of 19th century the French mathematician J.5) and Z π −π cos mx cos nxdx = 0. cos nx. Thus. Fourier was led to the discovery of certain trigonometric series during his research on heat conduction.1) Let us assume that f(x) is known on the interval (−π. 2.6) .B. cos mx with m 6= n.3) −π The coefficient an is determined similarly. It is easily verified that for integral values of m and n Z π sin mx cos nxdx = 0. . so that it can be integrated term by term from −π to π. 6. . . which now bear his name. (6. .4) where the missing terms are the products of the form sin mx . or of the form cos nx . . + an cos2 nx 2 (6. in the theory of sound. have become an essential part of the background of scientists.

±3π.8) and Eq. sin nx has the period. if n = 0 a0 = 1 π π f(x) cos nxdx = an Z π −π Z π cos 2 nxdx = an π −π (6. −2π. Upon setting x = ±π.1) f (x) = 2(sin x − 6. 2 Similarly.11) (6.9) (That is the reason for writing the constants term as 1 a0 rather than a 0 ). (6. (6.7) f(x) cos nxdx −π (6. (6. (6. . 2π. . 4π.1. . where p is a non zero constant. . (6. Example 6. in Eq. 72 (6. . where the sum of the series exhibits an abrupt jump from −π to π. . .1) which results when an and bn are determined is called the Fourier series of f (x).) 2 3 (6.12) sin 2x sin 3x + . (6.and hence integration of Eq.. we see that every term is zero.8). ±3π. .10) are called Euler-Fourier formulas and series in Eq. multiplying Eq. Now each term of Eq.4) yields Z so that an = In Eq. It remains to be seen what happens at the point x = ±π. .13).1) by sin x and integrating yields Z 1 π f (x) sin nxdx bn = π −π (6. For instance.13) has a period of 2π and hence the sum also has a period of 2π and the sum is equal to x on the interval −π < x < π and not on the whole interval −∞ < x < ∞.13) . (6.1 Represent the function f(x) = x by Fourier series over the interval (−π. π) Z 1 π an = x cos nxdx = 0 π −π Z π 1 x sin nxdx bn = π −π −2 cos nπ = n 2 = (−1)n+1 n Substituting in Eq. Any number p with this property is a period of f(x). Hence the sum is zero.1 PERIODIC FUNCTIONS A function f (x) is said to be periodic if f(x + p) = f(x) for all values of x.10) The formulas of Eq.8) Z π f(x)dx −π (6. . (6. . (6.

2 Find the Fourier series of the function defined by f(x) = 0 if f (x) = π if −π ≤x< 0 0≤x≤π (6.17) (6.) 2 1 3 5 (6.14) ∙Z 0 ¸ Z π 1 0. π +2 2 µ sin 2x sin 4x sin 6x + + +.18) The successive partial sums are y0 = Example 6. and a0 is called the fundamental or d − c term of the Fourier series.21) 2 sin 3x π π π .15) (6.19) 73 . n 1 − cos nπ 1 2 2 0 3 2 4 0 5 2 6 0 Determining bn by this table. π < x < 0 f(x) = π. 0 < x < π f(x) = is obtained. we obtain the required Fourier series f(x) = sin x sin 3x sin 5x π + 2( + + + . . 2 2 2 3 (6.. y 2 = + 2 sin x + etc.3 Find the Fourier series of the function defined by n −π. Example 6..16) (6.The term an cos nx + bn sin nx is sometimes called the nth harmonic. 2 4 4 ¶ (6. f or n ≥ 1 an = π 0 Z π 1 bn = π sin nxdx π 0 1 = (1 − cos nπ) n a0 = The factor (1 − cos nπ) assumes the following values as n increases. . y1 = + 2 sin x.dx + πdx = π π −π 0 Z π 1 π cos nxdx = 0.

although it is continuous and differentiable for every value of x.2 REMARKS ON CONVERGENCE In Eq. A function f(x) is said to be bounded if |f(x)| < M holds for some constant M and for all x under consideration. Example 6. (6. The Fourier series for 1 [f (x+) + f(x−)] 2 (6. sin x is bounded.24) 74 .23) converges at every value of x.1) each term has a period of 2π and hence if f(x) is to be represented by the sum.3 and Fig. (6. Let f(x) be defined for other values of x by the periodicity condition f (x + 2π) = f(x). is bounded. that is lim f (x0 +) 6= lim f (x0 −). 6. The function sin(1/x) has infinitely many maxima near x = 0 and the discontinuity at x = 0 is not simple. all its discontinuities are simple. 6. f(x) must also have a period of 2π. has only a finite number of maxima and minima and only a finite number of discontinuities. we shall suppose that f(x) is on the interval (−π.2). The function defined by the 1 f(x) = x2 sin( ) x 6= 0. f (0) = 0 x also has infinitely many maxima near x = 0.3 DIRICHILET’S THEOREM Suppose f(x) is defined on the interval (−π. but are unequal. even though the latter is well defined for every value of x.4.22) The simple discontinuity is used to describe the situation that arises when the function suffers a finite jump at a point x = x0 (Fig. 6. but the function f(x) = x−1 for x 6= 0. Analytically this means that two limiting values of f(x) as x approaches x0 from the right and left hand sides exist. That is f(x+) and f (x−) exist at every value of x. For example. π). It can be shown that if a bounded function has finite number of maxima and minima and only a finite number of discontinuities. The condition imposed on f(x) are called Dirichlet’s conditions after the mathematician Dirichlet who discovered the theorem.6. 6.1). and hence it converges to f(x) at points where f (x) is continuous. The behaviour of these two functions is illustrated graphically in Fig.4 Consider the Fourier series of a periodic function defined by f(x) = −π for − π < x < 0 f(x) = x for 0 < x < π (6. such as Eq. π) and outside this interval f (x) is determined by the periodicity condition f(x + 2π) = f(x) (6. (f (0) = ∞) is not. Whenever we consider a series.

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30) (6. cos x and x2 are even and x and sin x are odd...25) For n = 0 a0 = − π Similarly. the Fourier sine and cosine coefficients can be determined by inspection. (6. equality holds at all points of continuity.31) . 6. 76 (6.26) Therefore.. and =0 2 2 2 respectively.an = ∙Z 0 ¸ Z π 1 −π cos nxdx + x cos nxdx π −π 0 ∙ ¸ 1 cos π − 1 = π n2 (6. A function f(x) is said to be even if f(−x) = f(x) and the function f (x) is odd if f(−x) = −f(x) For example. the Fourier series is f(x) = − +3 sin x − 2 2 cos 3x 2 cos 5x π − cos x − − − . 8 1 3 5 7 as is seen by making substitution in Eq.4 EVEN AND ODD FUNCTIONS For many functions. At the points of discontinuity x = o and x = π. the series converges f (0+) + f (0−) π f(π+) + f (π−) = ..28) (6. since f (x) has been defined to be periodic..27) sin 2x 3 sin 3x sin 4x 3 sin 5x + − + −. Either condition leads to the interesting expansion π2 1 1 1 1 = 2 + 2 + 2 + 2 + .. 2 3 4 5 By Dirichlet’s theorem.27).29) (6. 2 bn = ∙Z 0 ¸ Z π 1 −π sin nxdx + x sin nxdx π −π 0 1 = [1 − 2 cos nπ] n (6. 4 π π 32 π 52 (6.

let f (x) be even. so that bn = Example 6. bn = 0 0 (6.32) and Z α f (x)dx = 0 if f(x) is odd −α (6. Then f (x) cos nx is the product of even functions. f(x) = x.33) Products of even and odd functions obey the rules (even) (even) = even (even) (odd) = odd (odd) (odd) = even The product of sin nx and cos mx is odd and Z α sin nx cos mxdx = 0 −α (6.34) Theorem 6.38) . the series has sine terms and the coefficients are given by bn = 2 π Z π f(x) sin nxdx. 77 1 π Z π f(x) sin nxdx = 0 −π (6.37) On the other hand f(x) sin x is an odd function. Therefore.6. −π < x < π. Fourier series has cosine terms only and the coefficients are given by an = 2 π Z π f(x) cos nxdx. defined in the interval −π < x < π is even.Also.35) If f(x) is odd. Z 1 π f(x) cos nxdx π −π Z π 2 f (x) cos nxdx = π 0 an = (6.5 Consider the function in Fig. Z α f(x)dx = 2 −α Z α f(x)dx if f (x) is even 0 (6.36) In order to see this. an = 0 0 (6.1: If f (x).5 where.

π) which satisfies the Dirichlet’s conditions can be expanded in a sine series and cosine series on 0 < x < π.6 (6.47) converge to the same function x when 0 ≤ x < π.42) and (6.(6.40) (6. .) − π < x ≤ π 2 3 4 bn = (6. f(x) = |x| or − π ≤ x ≤ π The function is even.6 and write its Fourier series. − π ≤ x < π − 2 π 12 32 52 (6..Since the function is odd.47) is the Fourier cosine series. .45) x cos nxdx ¯π Z π ∙0 ¸ x sin nx ¯ sin nx ¯ − dx = ¯ n n 0 0 2 n = 2 [(−1) − 1] nπ (6.46) |x| = ¸ ∙ 4 cos x cos 3x cos 5x π + + + .44) (6.42) is called Fourier sine series for x and Eq.42) Consider the function in Fig. To obtain the sine series. 6. series in Eqs. Any function f(x) defined in (0. (6.. the Fourier series reduces to a sine series Z 2 π x sin nxdx π 0 ∙ Z ¸ 2 x cos nx ¯π 1 π ¯ = − cos nxdx ¯ + π n n 0 0 ¯π ¸ ¯π ∙ ¯ 2 −x cos nx ¯ ¯ + sin nx ¯ = ¯ 2 π n n ¯0 0 2 = (−1)n+1 n sin 2x sin 3x sin 4x + − + . 78 . The first expansion of Eq.43) (6. we extend f(x) over the interval −π < x < 0 in such a way that the extended function is odd. (6.39) (6. The Fourier series for f(x) consists of sine terms only since f(x) is odd.47) Since |x| = x for x ≤ 0.41) x = 2(sin x − Example 6. hence an = = a0 = an = 1 π 2 π 2 π 2 π 2 π Z π Z−π π Z0 π Z0 π |x| cos nxdx x cos nxdx xdx = π (6.

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T ). To obtain a sine series. then corresponding coefficients are equal.5 EXTENSION OF INTERVAL The methods developed upto this point restrict the interval of expansion to (−π. ∙ ¸ ∞ ∞ X a0 X TZ = + an cos nz + bn sin nz π 2 n=1 n=1 πx T f (6.3) becomes 80 .. it is desired to develop f (x) in Fourier series that will be valid over a wider interval. one may get an expansion valid for all x. 22 − 1 4 − 1 6 − 1 when 0 < x < π. Since z = .. the series in Eq. That is just not a coincidence as shown by the following.7 Obtain a cosine series and also a sine series for sin x For the cosine series an = Z 2 π sinx cos nxdx π 0 2(1 + cos nπ) = n 6= 1 π(1 − n2 ) (6. hence sin x = 2 4 − π π µ ¶ cos 2x cos 4x cos 6x + 2 + 2 +. π). it converges to | sin x| rather than sin x when π < x < 0.49) For n = 1. Since the sum of the series is an even function.51) from −π ≤ z < π. the result of integration is zero.50) 0 Hence the Fourier sine series for sin x is sin x. the function f(T Z/π) can be developed in a Fourier series in Z. (6. To obtain an expansion valid on the interval (−T. By letting the length of the interval increase indefinitely. .Example 6.48) (6. T ) change the variable from x to T Z/π If f (x) satisfies the Dirichlet’s conditions on (−T.1 UNIQUENESS THEOREM: If two trignometric series of the form of Equation (6. 6.1) converge to the same sum for all values of x. 6. In many problems.4. an = 0 n > 2 Z π bn = 2 π sin x sin nxdx = 0 n > 2 1 n=1 (6.

56) (6.55) (6. and that the integral 81 . (6.8 Let 1 T T f(x) sin −T nπx dx T (6.dx + 1.dx = 1 2 −2 0 ∙Z 0 ¸ Z 2 1 nπx nπx dx + dx 0.58) f(x) = 1 3πx 1 5πx 1 2 + (sin πx + sin + sin + .51) ¸ Z ∙ 1 π TZ cos nz dz π −π π Z T 1 nπx = dx f (x) cos T −T T an = (6. Assume that f(x) satisfies the Dirichlet conditions in every interval (−T.57) (6. T ) and it is natural to enquire if a representation for arbitrary function on (−∞. cos 2 −2 2 2 0 ¯2 1 nπx ¯ ¯ =0 sin nπ 2 ¯0 ∙Z 0 ¸ Z 2 1 nπx nπx dx + dx 0.54) f (x) = 0 for − 2 < x < 0 f (x) = 1 for 0 < x < 2 a0 = an = = bn = = ∙Z 0 ¸ Z 2 1 0.59) Subject to Dirichlet conditions. . cos 1. the function can be chosen arbitrarily on the interval (−T . T ). sin 1.8) to Eq.f (x) = a0 X nπx X nπx an cos bn sin + + 2 T T n=1 n=1 ∞ ∞ (6. (6. which has many practical applications.52) By applying Eq. We shall see that such a representation is possible. sin 2 2 2 2 0 1 (1 − cos nπ) nπ (6. ∞) might be obtained by letting T → ∞.) 2 π 3 2 5 2 (6. .53) and Z bn = Example 6. The process leads to Fourier Integral Theorem. no matter how large.

63) can be written as ¸ ∞ ∙Z T 1X f (t) cos n∆ω(t − x)dt π n=2 −T where ∆ω = π T ¯ Z T ¯ 1 M f (t)dt¯ ≤ |f (t)|dt ≤ ¯ 2T 2T −T (6. R∞ |f (x)|dx is assumed to be convergent −∞ ¯ Z ¯ 1 ¯ ¯ 2T T −T which obviously tends to zero as T is allowed to increase indefinitely.62) Substituting these values of coefficients in Eq. Also. cos f(t)dt + (6. T ) is made large enough. For large values of T 82 . can be made as small as desired.M = Z ∞ −∞ |f(x)|dx (6. if the interval (−T .64) Moreover. As we have just seen f(x) is given by a0 X nπx X nπx + + an cos bn sin 2 T T n=1 n=1 ∞ ∞ f (x) = where 1 T Z T (6. (6.61) an = f (t) cos −T nπt 1 dt.61) 1 2T Z T ∞ Z 1X T nπ(t − x) dt f(t) cos T T −T n=1 f (x) = Since. Therefore. the quantity π/T which appears in the integrands of the sum.66) is very small. The sum suggests the definition of the definite integral of the function Z T F (ω) = −T f(t) cos ω(t − x)dt in which the values of the function F (ω) are calculated at the points n∆ω.63) −T nπx nπt nπx nπ(t − x) nπt cos + sin sin = cos T T T T T (6.60) converges.65) (6. the sum in Eq. bn = T T Z T f (t) sin −T nπt dt T (6. (6.

if the function f(x) satisfies the above conditions. f (t) sin ωt is even. the sum will approach the limit 1 π Z ∞ dω 0 Z ∞ −∞ f(t) cos ω(t − x)dt (6. then both the above integrals be used. the integral converges to zero at x = 0. Similarly. in 2 π Z ∞ 0 f (x) = dω Z ∞ 0 f(t) sin ωt sin ωxdt 83 . then f (t) cos ωt is an odd function. this fact is verified by setting x = 0. So that f (x) = 2 π Z ∞ 0 dω Z ∞ 0 f(t) sin ωt sin ωxdt (6. In particulars.72) when f(x) is odd. A similar argument shows that if f(x) is even.70) The foregoing discussion is heuristic and cannot be regarded as a rigorous proof. then Eq.71) If f(t) is odd. (6. the validity of the formula can be established rigorously. the Fourier 2 integeral also does. This formula assumes a simpler form if f (x) is even or an odd function.67) ∞ −∞ f (t) cos ω(t − x)dt (6. for an odd function. then 2 π Z ∞ 0 f(x) = dω Z ∞ 0 f (t) cos ωt cos ωxdt (6.63) can be written as f(x) = 1 π Z ∞ 0 dω Z ∞ −∞ f (t) cos ω(t − x)dt (6.69) If such is the case. Expanding the integrand of the integral: 1 π Z ∞ 0 dω ∙Z ∞ −∞ f(t) cos ωt cos ωxdt + Z ∞ −∞ ¸ f (t) sin ωt sin ωxdt (6.Z differ little from Z T −T f(t) cos ω(t − x)dt (6. Since the Fourier Series converges to 1 [f(x+) + f (x−)] at points of discontinuity. However.68) and it appears plausible that as T increases indefinitely.73) If f(x) is defined in (0. ∞).

. ω 6= 0 ω (6.79) .74) ∞ 0 π sin ω cos ωx dω = if 0 ≤ x < 1 ω 2 π = if x = 1 4 = 0 if x > 1 (6.77) Upon recalling the definition of f(x).Example 6.6 COMPLEX FOURIER SERIES . Then Z 1 f(t) cos ωtdt = 0 Z 1 cos ωtdt = 0 sin ω .75) We choose f (x) = 1 for 0 ≤ x < 1 and f (x) = 0 for x > 1.FOURIER TRANSFORM The Fourier series a0 X + (a n cos nx + bn sin nx) 2 n=1 ∞ (6. we see that the desired result is obtained for 0 ≤ x < 1.9 By f (x) = obtain the formula Z 2 π Z ∞ 0 f(t) cos ωt cos ωxdt (6.76) Z ∞ 0 π sin ω cos ωxdx = f(x) ω 2 (6.76) substituting in Eq. (6.78) f (x) = with an = Zπ 1 f(t) cos nt dt π −π Zπ 1 bn = f(t) sin nt dt π −π can be written with the aid of Euler formula e jµ = cos µ + j sin µ 84 (6. The fact that the integral is π when x = 1 follows from 4 f(1−) + f(1+) 1 = 2 2 6.

we can obtain the above formula for Cn .84) Z π eαt e−jnt dt −π (6. namely ∞ X f (x) = Cn e jnx (6. If the series ∞ X f (x) = Cn e jnx n=− ∞ is uniformly convergent.83) m= −∞ If we now integrate from −π to π the terms with m = n integrate to zero and the term with m 6= n give 2πC n giving Cn = Example 6.85) e( α− jn)t dt −π (6. π) Hence.81) and the limit is interpreted by taking the sum from −n to +n letting n → ∞. Replace x by t and the dummy index n by m. Thus.in an equivalent form. This can be shown as below.10 Consider the function f (x) = eαx on (−π.80) n=− ∞ The coefficients Cn are defined by the equation Cn = 1 2 Z π f(t)e−jnt dt −π (6. the index n runs through all positive and negative integral values including zero.86) 85 . so that ∞ X f(t) = Cn ejmt (6.82) m=−∞ Multiplying by e−jnt ∞ X f (t)e −jnt = Cm e j(m−n)t (6. 2πCn = Z π 1 2 Z π f (t)e −jnt dt −π (6.

Cn = eαπ − e −απ (−1)n 2π α − jn sinh α (−1)n (α + jn) = π α2 + n2 (6. 6.11 Consider the rectangular pulse train shown in Fig.7 and draw its amplitude spectra. ∞ sinh πα X (−1)n (α + jn)ejnx π α2 + n2 n=− ∞ eαx = (6. it is an even function Ae −jnωtdt − τ /2 Cn = 2 T Z τ /2 A cos nωtdt 0 = 2A sin nωt τ /2 2A sin nωτ /2 | = T nω 0 T nω Aτ sin cntf τ T 86 = f = ω 1 = 2π T . Therefore. 1 T Z τ /2 Cn = Since. The pulse width is τ and the period is T .87) Hence.88) Example 6.

sin cnt = Therefore.91) 1 2π Z ∞ −∞ e −jωt f(t)dt (6. 1 A→∞ 2π Z A 1 2π Z ∞ −∞ e −jω f (t)dt (6. It is one of the most powerful tools in modern analysis. the Fourier series is as follows f (t) = sin nt nt ∞ Aτ X sin(nf τ)e jnωt T n=− ∞ The amplitude spectrum then is drawn below Let us now write the Fourier Integral Theorem as 1 A→∞ 2π Z A f (x) = lim dω −A Z ∞ −∞ e jω (x−t) dt (6.92) is called the Fourier transform.90) f(t) = lim The transform T defined by T (f) = e jωtg(ω)dω −A (6.89) when f(x) satisfies the Dirichlet conditions g(ω) = then. 87 .

θn (x) = sin nx is orthogonal on (0.98) For example. so does the other. π) because 88 .96) are equal. (6. (6. One can write g(t) in Eq.Although. and suppose Z ∞ Z ∞ |f (t)| 2 dt or |g(ω)|dω (6.1 PLACHEREL’S THEOREM: Let f (t) and g(ω) be integrable on every finite interval. and the two integrals of Eq. It is said that g A(t) converges in mean to g(t) and we write g(t) = lim gA(t) A→∞ (6. 6.6. This is in the sense of ordinary Reimann integral.93) If it is true that lim Z ∞ −∞ A→∞ |g(t) − gA (t)|2 dt = 0 Z A (6. A more symmetric theory can be based on a type of convergence known as mean convergence.97) holds in the sense of mean convergence.96) −∞ −∞ is finite. the formulas of Eq. (6. then if either of the equations Z ∞ 1 g(ω) = f (t)e −jωt dt 2π −∞ Z ∞ 1 f (t) = g(ω)ejωt dω 2π −∞ (6.94) holds with gA (t) replaced by the integral on the right of Eq.91) and Eq. if it is stated that the equation holds in the sense of mean convergence.94) As an illustration g(t) = lim A→∞ e− jω tf (t)dt (6. (6. b) if Z b θ m (x)θn (x)dx = 0 for m 6= n a 6= 0 for m = n (6.95). Let gA (t) be an integrable function of t on each finite interval for each value of parameter A.95) as an integral from −∞ to +∞. (6. the conditions on the functions f and g are quite different.7 ORTHOGONAL FUNCTIONS A sequence of functions θn (x) is said to be orthogonal on the interval (a. θ 2 (x) = sin 2x. the sequence θ 1 (x) = sin x.92) are similar. 6. (6.95) −A means that Eq.

. the cross-product terms disappear. . b) and suppose that another function f (x) is to be expanded in the form f(x) = c1 φ1 (x) + c2 φ2 (x) + . . 0 Z 2π sin 2 nxdx = π. π) The formula for Fourier coefficients is specially simple if the integeral has the value of 1 for m = n. (π)−1 /2 sin x. If Z In other words Z For example.103) . (π)−1 /2 cos x. (π)−1/2 sin nx. but the square of each function gives 1 when integrated from 0 to 2π. 2π) though not on (0. and hence 89 (6. (π)−1 /2 cos nx The product of two different functions in this set gives zero. The function θn (x) are then said to be normalized and {θ(x)} is called an orthonormal set. sin 2x.102) (6. .100) b a φm (x)φn (x)dx = 0 for m 6= n = 1 for m = n (6. sin x. . + cn (φn (x))2 If we formally integrate from a to b. . + cn φn (x) To determine the coefficient cn . 0 Z 2π cos2 nxdx = π 0 for n ≤ 1 and the orthonormal set is (2π)− 1/2 . since Z 2π b [θn (x)]2 dx = 1 a (6.dx = 2π.99) The sequence 1. cos 2x is orthogonal on (0.101) 1.Z π θ m (x)θn (x)dx = 0 Z π 0 sin mx sin nxdx = 0 for m 6= n = π for m = n 2 (6. Let {φn (x)} be an orthonormal set of functions on (a. cos x. . . we multiply by φn (x) getting f (x)φn (x) = c1 φ1 (x)φn (x) + c2 φ2 (x)φn (x) + .

the convergence of Rb Rb f 2 dx and a φ2 dx is required. . the quantity |f(x) − pn (x)| or [f (x) − pn (x)] 2 (6. (6. If either expression of Eq.107) involves an integration that is not present in Eq. (6.106) for Fourier series. These measures of error are appropriate for discussing convergence at any fixed point x. 6. The sequence pn (x) converges to f(x) whenever the expression of Eq. the coefficients cn must be given by Eq. the result is precisely the mean value of the corresponding expression of Eq. (6. Even though Eq. The terminology is appropriate because if the integrals of Eq. (6. a ≤ x ≤ b.104) The term to term integration is justified when the series is uniformly convergent and the functions are continuous. the sequence p(x) is said to converge in mean to f(x) and the mean convergence is used. But it is often useful to have a measure of error which applies simultaneously to a whole interval of x values. it is much easier to discuss the mean square error and the corresponding mean convergence then the ordinary convergence.105) gives a measure of the error in the approximation. Such measure is easily found if we integrate Eq. n a Let {φn (x)} be a set of orthonormal functions on a ≤ x ≤ b.Z b f (x)φ(x)dx = a Z b Cn [φn (x)] dx a 2 (6. The foregoing procedure shows that if f(x) has an expansion of the desired type. (6. (6.107) These expressions are called mean error and mean-square error respectively. so that as in the preceding section Z b a φn (x)φm (x)dx = 0 for m 6= n = 1 for m = n 90 (6. the coefficients cn are called the Fourier coefficients of f (x) with respect to {φn (x)} and the resulting series f(x) = c1 φ1 (x) + c2 φ2 (x) + .107) approaches zero at n → ∞ .108) .108) from a to b.106) (6. Z b a |f (x) − pn (x)|dx or Z b a [f (x) − pn (x)] dx 2 (6. (6.106). (6. If integrals are improper. In the following discussion.8 MEAN CONVERGENCE OF FOURIER SERIES If we try to approximate a function f(x) by another function pn (x). + cn φn (x) is called the Fourier series with respect to {φn (x)}. we use f and φn as abbreviation for f(x) and φn (x) respectively and assume that f and φn are integrable on a < x < b.106) approaches zero as n → ∞. .(104) is called Euler-Fourier formula.107) are multiplied by 1/(b − a).

+ an φn (x) in such a way that mean square error of Eq.113) (6. + an φn )] dx = min 2 (6. . + an cn a (6. Z Z Z b b b E= a f 2 dx −2 (a1 φ 1 + a2 φ2 + . . + an φn )(a1 φ 1 + a2 φ2 + .116) for the mean square error in the approximation.We seek to approximate f(x) by a linear combination of φ(x).112) The third integral in Eq.116) is also equal to Z b n n X X E= f 2 dx − c2 + (a k − ck )2 k a k=1 k=1 (6.110) yields. . . Hence Eq. + a2 φ2 dx 1 1 2 2 n n a 2 1 Where the second group of terms involves cross products φ iφj with i 6= j and such terms integrate to zero. . . (6. ck = The second integral in Eq. E= Z b a [f − (a 1 φ1 + a2 φ2 + . Z b n X k=1 n X k=1 E= a f 2 dx − 2 ak ck + a2 k (6. . we see that Eq. (6. (6. In as much as −2a kck + a2 = −c2 + (ak − ck )2 k k The error E in Eq.110) can be written as Z b (a 1 φ1 + a2 φ2 + .110) a If the Fourier coefficients of f relative to φ k are denoted by ck . (6. + a n φn )dx (6.109) yields. .109) Upon expanding the term in brackets. + an φn )fdx + a (a1 φ1 + a2 φ2 + . .110) is Z b Z b φk fdx a (6. (6. + a2 2 n Z a2 φ2 + a2 φ2 + . . + an φ n )dx a b (6. . . pn (x) = a 1 φ1 (x) + a2 φ2 (x) + . . .114) (6. . . .117) 91 . .111) (a1 φ 1 + a2 φ2 + . . + an φ n )fdx = a1 c1 + a2 c2 + .115) = = a + a2 + . .107) is minimum. (6.

Corollary 1: The partial sum of the Fourier Series c1 φ 1 + c2 φ 2 + . Letting n → ∞ in Eq. + an φn upon setting a ak = ck in Eq. we obtain a number of interesting and significant theorems. Rb Corollary 3: The Fourier coefficients cn = a fφ n dx tends to zero as n → ∞. + cn φ n .Theorem 6. From the two expressions of Eq. the terms (ak − ck )2 in Eq. For applications. we see that the minimum value of the error is min.1179). then the series k=1 c2 converges and satisfies the Bessel inequality k n X k=1 c2 ≤ k Z b [f(x)] 2 dx a (6. . the error approaches zero for some choice of a0 s only. In the first place. is not negative.109) shows that E ≥ 0 because the integrand in Eq. in which case they are zero.119) Now.109) being a square. it is important to know whether or not the mean square error approaches zero as n → ∞. the expression of Eq. Evidently. "Z or b n X k= 1 a f dx − n X k= 1 2 c 2 k # ≥ 0. ck = Z b f φk dx a (6.118) gives a smaller mean square error Rb (f − φ k )2 dx then is given by any linear combination a1 φ1 + a2 φ2 + .117).117) where ck are the Fourier coefficients of f relative to φk . (6. (6.117) are positive unless ak = ck .109) can be written in the form of Eq (6.E = Z b a f 2 dx − n X k=1 c2 k (6.120) Upon letting n → ∞. we get Parseval equality 92 . . Hence the choice of ak that make E minimum is obvious by ak = ck and we have the following. (6. Since E ≥ 0 for all choices of a k .if the minimum error in Eq.117) does so. . (6. Rb Corollary 2: If ck = a fφk dx are the Fourier coefficients of f relative to the orthonormal P∞ set φ . .121) Since the general term of a convergent series must approach zero. Hence. the mean square error of Eq (6.117). c2 ≥ k Z b f 2 dx a (6. we obtain by the principle of monotone convergence. (6. we deduce the following from Corollary 2.112) and Eq. (6. (6.2: If {φn (x)} is a set of orthonormal functions. (6. it is clear that minimum of E (which arises when ak = ck is also greater or equal to zero.

That is.9 POWER IN A SIGNAL Consider two voltage sources connected in series across a 1 ohm resistance. 6.124) [f(x)]2 dx = 0 a (6. 2. 3 .122) Corollary 4: If f is approximated by the partial sum of its Fourier series. v3 are not coplanar.Z as the condition for zero error. . Thus a set of vectors v1 . hence we have. The set of vectors v1 . . v2 . The notions of closure and completeness have simple analogs in the elementary theory of vectors.123) holds.126) for some choice of constants ck .125) at once. Corollary 5: Every closed set is complete. This converse. (6. v2 . however requires a more general integral than the Reimann. sin nx is closed on 0 < x < 2π.127) In this case. v. The converse is also true. that is the set is complete if the condition. whenever Eq. the mean square error approaches zero as n → ∞ if and only if Bessel inequality becomes Parseval’s equality ∞ X k=1 c2 = k Z b [f(x)] 2 dx a (6. . the set {φ n (x)} is said to be closed. Eq. the Fourier series converges to f in the mean square sense if and only if Eq. it is obvious that closure and completeness are equivalent. 3. . (6. (6. 2. . for both conditions simply state that the three vectors v1 . a (6. A closed set then is a set which can be used for mean square approximation of arbitrary functions. The generalized integral known is Lebesgue integral. (6. If this happens for every choice of f.vk = 0 k = 1. These two voltages do not make a periodic function. the set is complete if ck = implies that Z b Z b f (x)φ(x)dx = 0 for k = 1.123) holds. (6. A set {φn (x)} is said to be complete if there is no non trivial function f(x) which is orthogonal to all the φ0n s.123) In other words. v2 . 93 . v3 is said to be complete if there is no nontrivial vector orthogonal to all of them. V = c1 v1 + c2 v2 + c3 v3 (6.125) Now. b a f 2 dx − ∞ X k=1 2 Ck = 0 (6. It can be shown that the set of trigonometric functions cos nx. v 3 is said to be closed if every vector V can be written in the form. Let one source have an emf of 10 cos 2πt and other an emf of 5 cos 20t.124) yields Eq.

130) and root mean square value of f(t) ∞ X r.129) From Eq.5 cos 40t + 50 cos(2π + 20)t + 50 cos(2π − 20)t (6.s. (6.If the power dissipated in the resistance at any moment is to be calculated. = n=−∞ |cn |2 (6.129). (6.131) are for two sided spectrum. (6.132) The Fourier series for the rectangular pulse train in Example 6.128) (6.129).5 is the average power if 10/πH z source acted alone. For the positive frequency line spectrum P = c2 + 0 ∞ ∞ X 1 X 2 |2cn |2 = c2 + 2 cn 0 2 n=−∞ n= 1 (6. The instantaneous power is given by Eq. 6.131) The expressions of Eq.5 + 50 cos 4πt + 12. The total average power when both sources are present is the sum of the averages for both sources acting alone.130) and Eq.1 AVERAGE POWER IN A SIGNAL Applying the Parseval equality ∞ X ∞ X −∞ P av = cn c−n = n= −∞ |cn | 2 (6. (6.11 was ∞ Aτ X sin cnfejnωt T n=− ∞ f (t) = where cn = Aτ sin cnfτ T c0 = A T The ratio τ /T is called the duty cycle ‘d0 94 .9. we have p(t) = v2 (t) = v 2 (t) = (10 cos 2πt + 5 cos 20t)2 R = 100 cos2 2πt + 100 cos 2πt cos 20t + 25 cos2 20t = 50 + 12.m. it is clear that 50 is the average power that would be dissipated in the load if 1Hz source acted alone and 12.

. the series can be written in cosine terms. Therefore. 95 . f (t) = c1 cos πt + c2 cos 2πt + c3 cos 3πt + . cn = dn A sin cnd Then average power ∞ X P av = (dA)2 sin c2 nd n=−∞ Example 6. 6.13 The triangular wave is shown in Fig. .10.thus. since the wave has no average value and ∙ sin πn 2 πn 2 cn = ¸2 f or n 6= 0 Since all ck are real. 6.9. The Fourier series is ∞ 1 X cn ejπ nt T n=− ∞ f (t) = we can also define cn as cn = Z T f (t)e 0 −j 2πnt T dt where c0 = 0. Example 6. Draw its amplitude spectrum and write the expression for average power 1 T = The average power Pav is ∞ X cn = Z τ /2 A cos ωc t cos nωtdt Aτ [sin c(fc − nf)τ + sin c(fc + nf)τ ] 2T − τ /2 P av = c2 n n=− ∞ and the amplitude spectrum is given in Fig. Draw the power spectrum of the function.11 along with its Fourier series. T = 2. 6.12 Consider the train of sinusoidal pulses in Fig.

.

. . 6. but we can define: ∞ u(t) X cn e j2πnf t T n=−∞ fin (t) = 97 .Moreover.) (cos πt + cos 3πt + π2 9 25 49 The power spectrum (also called line spectrum) is obtained by P avn = |cn | 2 16 4 = = 4 4 watts T2 (nπ)4 (2)2 πn The line spectrum is plotted in Fig. 3.10 PERIODIC SIGNAL AND LINEAR SYSTEMS If the input to a stable linear network or system is periodic. Then the sinusoidal form of the Fourier series. Strictly speaking fin (t) cannot be periodic if it is to have a Laplace transform. f (t) = 1 1 1 4 cos 5πt + cos 7πt + . 5. sin 2 (nπ/2) is zero when n is even and unity when n is odd. . the steady state output signal is also periodic with the same period.12.6. . cn .133) where H(s) is the transfer function of the system. the transform of the output signal is related to the transform of the input signal by the equation. F0 (s) = H(s)Fin (s) (6. therefore can be 4 written as cn = (π n)2 for n = 1. That this is true can be easily demonstrated by use of transfer functions. . If the system is linear.

cn = as the input signal.134) with H(s) the transfer function of a linear system ∞ 1 X cn H(s) T n=−∞ (s − j2πnf) F0 (s) = (6.136) and (6. 6. then ∞ 1 X c H (j2πnf )ej2π nf t T n=−∞ n f0 (t) = (6.135) F0 (s) will have poles in the left half plane because of the poles of H(s). The steady state effect of a filter on a signal can be seen if we compare the power spectrum of the signal with the frequency response of the filter.14.137) Eqs. The power spectrum of the output signal is given by P avn = |cn | 2 H(j2πnf)| 2 T2 (6. 6. These will lead to transient terms.136) and the only effect the system has on the series is to alter the amount of each frequency by the transfer function evaluated at that frequency.14 The rectangular wave of Fig. Its transform is Z T f (t)e 0 −j 2πnt T dt Fin (s) = ∞ 1 X cn T s − j2πnf n=− ∞ (6. Example 6. If we wish only the inverse transform of the steady state. The Laplace transform of one period of input signal is P (s) = e sT /4 − e−sT /4 s 98 .137) represent the principal reasons for the use of Fourier series in signal analysis. Find the Fourier series of the input and output signals and their spectra.13 is the input signal (the current is ) of the RLC tank circuit in Fig. we need only the inverse transform of the j-axis poles. If f0 (t) is the periodic portion only of the inverse transform. Multiplication of the two will produce the spectrum of the output signal.(6.

.

300. because frequencies other than 900 Hz get through. . 37. . Substitution of even harmonics (n even) will yield zero. we have cn = e jπ n/2 − e −jπn/2 2π jn T sin πn 2 =T nπ If we divide by T . a sharp resonance will be seen at f = 900Hz.Setting s = j2πn T . 80. 700. 7. 193. 3.7x10− 3 so the roots of the denominator are very nearly 2 s= 1 1 (− ± j) LC 20 = 5. 500. making the d − c term 1 s + ( ) + LC 1. At this frequency H(s) = R = 1000. The transfer function of the network is H(s) = s c s RC . 1000 and 240 respectively.65x103 (−0. 1100.5. Since the fundamental frequency of the 1 periodic wave is T = 100Hz. sin nπ = 2 (n = 0) = 1/2. Hz. 5. we obtain 11. sinx = x for small n. The circuit is not an ideal band pass filter.05 + j) If the frequency response versus frequency is plotted.77x10− 6 c Q 0 = R = 1000 = 10 L 17.5. If the numerator and denominator of H(s) are multiplied by H(s) = and with s= R 1 + ( RC )(s2 + s RC s . . the Fourier series of the input signal is ∞ X sin nπ 2 ej2π nt/T nπ n= −∞ nπ 2 is (t) = Note that for small angles. 900. 100 . H(s) can be written 1 LC ) j2πn = j 200πn T 1000 1 + ( j10 )(n − 8n1 ) 9 H(j2000πn) = substituting n = 1. 9. so the input signal consists only of the frequencies 100. but it certainly shows a preference for 900 H z.25. 11 and computing only the magnitude of H(s). the response of the network will be large at the ninth harmonic.

since R = 1000 and P = v2 /R. The dashed line in Fig. 6.25)2 = 12.16 shows the power spectrum of the output signal. We have: P av3 = 15. Note that it has no d − c term and that the line at n = 1 is 1/π 2 times the square of the magnitude of H at n = 1.7 cos 1800 πt in the steady state.15 is the magnitude of H(s) squared. the total is 1429. 6. Actually. .7 volts. that is (11.7 watts per ohm (the actual power is one thousandth of this. something less than this value is in the ninth harmonic. P av5 = 26. and the assumption that only the ninth harmonic is passed leads to the result that v(t) = 70. 9π The RLC is approximately a band pass filter. the output voltage should be nearly a 900Hz sinusoidal with average peak amplitude of 2x100 0 = 70.8 π2 calculating the others in a similar way. Fig. 6. . 101 .3.5 percent of the power is in the ninth harmonic.The Fourier series of the output is 1000 sin nπ 2 i .1.7. P av11 = 48. since the power in 13th.8.e 1200nπ t f0 (t) = v0 t = h 1 + ( j10 (n − 891 ) πn 9) The power spectrum of the input signal is simply a set of lines with height 1/4 at f = 0 and 1/(πn)2 at the odd harmonics. P av7 = 76. and so about 87. This is shown in Fig. . Since the output is a voltage and ninth harmonic is dominant. If the sum of the powers in these harmonics is calculated.15. P av9 = 1250. harmonics would have to be calculated to obtain the total output power. 15th.

PROBLEMS 6.1 Evaluate Rπ
−π

cos mx cos nxdx for integral m and n by use of the identity. 2 cos A cos B = cos(A + B) + cos(A − B) ½

6.2 Find the Fourier series for f (x) if f(x) = π, 0, for −π < x < for π < xπ 2
π 2

6.3 Find the Fourier series for the function defined by f(x) = 6.4 If f(x) = n 0, sin x, n −x, 0, for −π < x < 0 for 0 < xπ for −π < x < 0 for 0 < xπ

Show that the corresponding Fourier series is
∞ ∞ 2 X cos(2n − 1) X (−1)n sin nx π − + 2 4 π n=1 (2n − 1) n n=1

6.5 Classify the following functions as even, odd or neither x2 , x sin x, 6.6 Show that if f(x) = then f (x) = ∙ ¸ 2 cos 2x cos 6x cos 10x π − + + + ... 4 π 12 32 52 x3 cos nx, x4 , ex , (x2 )(sin x) 2 ½
π 2

x, π − x,

for 0 < x < for π < xπ 2

6.7 If f (x) is an odd function on (−T , T ) show that the Fourier series takes the form
∞ X n= 1

f(x) =

bn sin

nπt 2 , bn = T T

Z

T

f(x) sin

0

nπx dx T

6.8 Find the Fourier series for the following function: n for 0 < x < 2 f (x) = 8, −8, for 2 < x < 4 6.9 Write down the Fourier series for the waveforms shown in Figs. 6.17 (a)-(b)
103

6.10 For a one port network, it is given that i = 10 cos t + 5 cos(2t − 45o ) v = 2 cos(t + 45o ) + cos(2t + 45o ) + cos(3t − 60o ) a) what is the average power to the network. b) Plot the power spectrum. 6.11 By using the following equations e jµ = cos µ + j sin µ 1 2π Z
π

cn =

f (t)e −jnt dt
−π

show that, 2cn = an − jbn , 2c0 = a0 , 2c− n = an + jbn . 6.12 Determine whether f (t) is periodic. If it is, find its period and its fundamental frequency. Whether it is periodic or not, write the function in the exponential form and list all frequencies contained within the function. f (t) = 5 + 7 cos(20πt + 35o ) + 2 cos(200πt − 30o ) 6.13 The current source in the circuit shown in Fig. 6.18 is a square wave whose Fourier series is
∞ X

is (t) = with cn = 0 for n even, and

cn ej2π nt/T

n=− ∞

nπ sin cn = A nπ2 2 for n odd a) Sketch i s(t), v0 (t). b) Find the Fourier series of v0 (t). c) Write the first five nonzero terms of cosine series for v 0 (t). d) Calculate Pav. for v0 (t) if R = 100. Plot the power spectrum. e) Since the square wave can be thought as successions of steps, the steady state term v0 (t) must be a succession of step responses. Without using the Laplace transform, determine the waveshape of steady state.
105

.

.

Such signals may be strictly time limited.3) Eqs. music and audio signals) have both beginnings and ends.CHAPTER .5) remains finite as P → 0 and T → ∞. Since time limited signals must have zero averages.VII THE FOURIER TRANSFORMS INTRODUCTION In the preceding chapter on Fourier series.(7.5) The integral in Eq.4) P =< f 2 (t) >= lim 1 T →∞ T Z f 2 (t)dt 0 (7. since all the signals (speeches. so f (t) → 0 as t → ∞.3) are called the Fourier transform pairs. or if f(t) is square integrable over all time.2) and (7. that is lim T →∞ Z T −T |f(t)| 2 dt < ∞ (7. therefore.1) then the frequency domain description is provided by the transforms. signal transmission and filtering. we have shown that the period can be extended for non periodic signals and the resulting equations are called Fourier Transform pairs. the average of the signal is defined as 1 T →∞ T Z T < f(t) >= lim and power (t)dt 0 T (7. The practicality of the use of Fourier transform is validated by the fact that no practical signal is mathematically periodic. when averaged over infinite time. F (f ) = F[f (t)] F (f) = and f(t) = Z Z ∞ −∞ ∞ −∞ f (t)e−jωt dt (7.(7. By definition. Assuming that f(t) is applied to a one ohm resistor 108 .2) F (f)ejωt df (7. total energy E is the integral of instantaneous power. average power is. These transform pairs are extremely useful in dealing with the electromagnetic radiation. so f (t) is identically zero outside a specified interval or asymptotically time limited. not useful and we turn to energy.1 AVERAGE VALUE AND ENERGY IN A NON-PERIODIC SIGNAL Since the transforms are non-periodic functions. 7.

Moreover.10) S(f) is positive real. the frequency component represented by the interval F (f1 )(ejω1t ∆f + F (−f1 )ej(−ω 1) t∆f = 2F (f1 )∆f cos[ω1 t + ArgF (f1 )] This interval contains energy approximately equal to 109 .9) Eq. CONTINUOUS SPECTRA Consider a narrow frequency interval ∆f central at f1 that is f1 − 1/2∆f < |f | < f1 + 1/2∆f and suppose that this interval includes the mth harmonic of a periodic signal f1 = mf0 . By like reasoning |F (f)|2 is the density of energy in the frequency domain. average power is 2|cm |2 For a non periodic signal.9) is called Raleigh’s Energy Theorem. The frequency component of the periodic signal cm ejω1t + c−m ej( −ω 1)t = 2|cm |cos(ω1 t + φm ) so that. Z ∞ E= S(f)df −∞ Z ∞ =2 S(f )df 0 (7.8) F (f)F (−f)df |F (f)| 2 df (7.2 LINE SPECTRA VS. The total energy is therefore.7) (7.6) E= = = = f (t) Z−∞ ∞ Z−∞ ∞ Z−∞ ∞ −∞ F (f)e df dt − ∙Z ∞ ¸ ∞ F (f) f (t)e −(−jωt) dt df −∞ ∙Z ∞ jωt ¸ (7. Define S(f) as the energy spectral density S(f) = |F (f)| 2 (7.E= Using Parseval’s theorem Z ∞ Z ∞ −∞ f 2 (t)dt (7. If f (t) is a voltage waveform. F (f) is hermitian and S(ω) is even function of frequency. the F (f ) has dimensions per unit frequency and describes the distribution or density of the signal voltage in frequency. if f(t) is real. (7.11) 7.

14) 110 .12) (7. F (f) = Z τ /2 Ae−jωt dt −τ /2 sin ωτ 2 ωτ 2 = Aτ ωτ 2 ωτ S(f) = A2 τ 2 sin c2 2 = Aτ sin c The graph of F (f ) and S(f) is given in Fig. 1/τ can be taken as the measure of spectral width.1 whose amplitude is A between −τ /2 and τ/2 and zero otherwise.2 and Fig.13) In this example. (7. a line spectrum represents a signal that can be constructed from a sum of discrete frequency components and the signal power is concentrated at specific frequencies. 7. 7.2|F (f1 )|2 ∆f Therefore.1 Consider the time limited pulse of Fig.3. Draw its amplitude and line spectrum. this phenomenon is ”reciprocal spreading”. Using energy spectral density: τ E=2 S(f)df Z 1 /τ sin c2 fτ df = 2(Aτ )2 0 0 Z 1/τ (7. Now. Let us see the percentage of total energy contained in |f| < 1 . Example 7. 7. if the pulse width is increased and vice versa. On the other hand a continuous spectrum represents a signal that is constructed by integrating over a continuum of frequency components and signal energy is distributed continuously in frequency.

o nR t f(x)dx is 7. the Fourier transform of −∞ F (f)/(j2πf).3.5 If F (f) is the Fourier transform of f(t). 111 .92E Thus over 90 percent of signal energy is contained in |f| < Example 7. 7.2 If G(f ) and F (f) are Fourier transforms respectively of g(t) and f (t).15) −∞ = 2A = Ae 0 −π (f τ )2 Z e−π(t/τ)2 cos ωtdt (7.3 If F (f) is the Fourier transform of f(t). the Fourier transform of e j2πf 0t f(t) is F (f −f0 ).6 If F (f ) is Fourier transform of f(t).3. αF (f ) is a Fourier transform of αf(t). This is complex translation theorem of Laplace transforms with s = 2jπf0 . where t0 may be positive or negative real.2 Find and draw the amplitude spectrum of a Gaussain pulse of Fig. the Fourier transform of [g(t) + f (t)] is [G(f ) + F (f )] 7.5. 7. n ≥ 1 as f becomes large.3 FOURIER TRANSFORM THEOREMS Almost all Laplace transform theorems are also Fourier transform theorems. Actually F (f ) can approach a constant as f becomes infinite.16) is a Gaussian pulse in frequency as is shown in Fig.1 Multiplication by a scalar : If F (f ) is a Fourier transform of f(t).3. 7. 7. f0 may be positive or negative real.92A2 τ 2 = 0.3.4.16) Eq. the Fourier transform of f(t − t0 ) is e−j2 πf t0 F (f). provided this division by f does not produce a pole at f = 0. (7. f (t) = Ae− π(t/τ ) Z 2 1 τ F (f) = ∞ Ae −π( t/τ ) ∞ 2 cos ωtdt (7. 7.3. 7. Those that are the same will not be proved.4 Real Translation: If F (f ) is the Fourier transform of f (t). 7.3.= 0. the Fourier transform of df (t)/dt is j2πfF (f) Note that this assumes that F (f) behaves like 1/f n .

.

.

F (f /a) is the transform of f(at) 7. the Fourier transform of tf(t) is (j/2π)dF (f)/df .3 a) Since F[p(t)] = A sin πf b πf Where p(t) is a rectangular pulse of height A and width b and centered on the origin. Example 7.This theorem is peculiar to the Fourier transform and for the first time confuses the use of capital letters for transforms and lower case functions of time.8 If F (f) is the Fourier transform of f (t).Thus F (f ) must vanish at f = 0 at least as rapidly as f. Proof: We know that G(f) = and Z ∞ −∞ Z ∞ −∞ g(x)e− j2πf xdx (7. b) Since F [e−at u(t)] = 114 1 j 2πf + a .18) t is replaced by −f g(−f) = Z G(y)e −j2π f y dy which is seen to be the definition of G(t).3.3. Thus the Fourier transform of A sin(πbt)/πt is A for |f| < b and 0 for |f| > b.3. (7. g(−f ) is the Fourier transform of G(t). 7.9 If G(f) is the Fourier transform of g(t). More exactly lim | f →0 F (f ) |<∞ f 7. ∙ ¸ A sin πbt = p(−f) = p(f ) πt since the pulse p(t) is symmetric about t = 0 axis.17) g(t) = G(y)ej2π ty dy (7.18) If in Eq. replacing f by t in the transform and t by −f in p(t) gives.7 If F (f) is the Fourier transform of f (t).

dx = −dy. For example.3. the conjugate F (f).then.19) F (−f) = −∞ But Eq.19) is. the Fourier transform of eαt u(−t) . the Fourier transform of f (−t).10 If F (f) is the Fourier transform of f(t). Example 7. Proof: Since F (f) = then F (−f ) = Z Z ∞ −∞ ∞ −∞ f (x)e−j2πf x dx f(x)e j2πf xdx If the dummy variable of integration is changed to y = −x. F ∗ (f ). by definition. y = +∞.4 a) Since the Fourier transform of e −αtu(t) is 1 must be (− j2 πf +α) b) The Fourier transform of eαt cos tβ u(t) is j2πf + α s+α | = (s + α)2 + β 2 s=j2π f (j2πf + α)2 + β2 and the transform of e αt cos(−βt)u(−t) = e αt cos β(t)u(−t) is simply 115 1 j2π f +α . in the last pair above. Note that if f (t) is real. Z −∞ f(−y)e−j2π f y (−dy) F (−f) = ∞ The sign of the last integral can be changed if limits of integration are reversed. the Fourier transform of f(−t) is F (−f ). so Z ∞ f (−y)e −j2π f y dy (7. For x = −∞. F ∙ ¸ 1 = eaf u(−f) j2πt + a The theorem actually works both ways since either t is replaced by −f and f by t or t is replaced by f and f by −t. (7. put t for f on the right and obtain eatu(−t). For x = ∞. y = −∞ . is equal to F (−f). Now place −f for t on the left and get the pair F [e at u(−t)] = 1 −j2πf + a 7.

so f(t) = f(t)u(t) + f(−t)u(−t) and F (f) = F 0 (f ) + F 0 (−f ) Clearly. f(t) = f(t)u(t) +f (t)u(−t) since u(t)+ u(−t) = 1. so F (f ) = 2Re[F 0 (f)] 7. the transform f(−t)u(−t) is F (−f). But. so F (f ) = F 00 (f) − F 00∗ (f ) = 2jIm[F 00 (f )] Again there follows a corollary: If f (t) is an odd function of time and F (f ) is its transform. F 0(−f) = F ∗ (f) by Theorem 10. Proof: Evidently f (t) = f (t)u(t) + f (t)u(−t) and because f(t) is odd f(t) = f (t)u(t) − f (−t)u(−t) If F 0 0(f ) is the transform of f(t)u(t). if it exists.12 If f(t) is real and odd function of time that is. Then by Theorem 10. Proof: Let the Fourier transform of f (t)u(t) be F (f ).3. if f(t) = f (−t) and if f(t) is transformable. The even property of f (t) permits the second f (t) in the last equation to be replaced by f(−t). the equation does not change. its Fourier transform F (f ) is real and is an even function of f .11 If f(t) is a real and even function of t that is.−j2πf + α (−j 2πf + α)2 + β2 7.3. the transform of f(t) is F (f ) = F 00 (f) − F 00 (−f ) which is seen to be odd. Then F (f) = F 0 (f ) + F ∗ (f) But the sum of any complex number and its conjugate is twice the real part of the number.. the transform of f(t). is imaginary and an odd function of f .if f (t) = −f (−t). The fact that f (t)u(t) is real means that F 00∗ (f ) − F 00 (−f ). F (f) is even because if f is replaced by −f. But since f(t)u(t) is real. the imaginary part of the transform of f(t)u(t) is −jF (f )/2 116 0 .

4 SUMMARY OF FOURIER TRANSFORM THEOREMS All the theorems in the preceding section are summarized by the following equations.20) fe (t) = and f0 (t) = (7.30) provided that F (f ) is bounded for f = 0 117 .23) (7.26) (7.27) (7. Let f(t) and g(t) be transformable functions with transforms F (f) and G(f) respectively. since for f(t) neither even nor odd. Then F[af(t)] = aF (f) F[f (t) + g(t)] = F (f ) + G(f ) ¤ £ F ej2π f0t ft = F (f − f0 ) F[f(t − t0 )] = ej2π t0f F (f) ∙ ¸ df(t) = j2πf F (f ) dt ¸ F (f) f(x)dx) = j2πf ∞ t (7.22) Then Theorems 11 and 12 imply that if F (f) is the transform of f (t) F[fe (t)] = Re[F (f)] F[f0 (t)] = jIm[F (f)] 7. f(t) = fe (t) + f0 (t) with f(t) + f (−t) 2 f(t) − f(−t) 2 (7.21) (7.28) F (7.25) (7.24) (7.29) provided that fF (f) is bounded as f → ∞ F ∙Z (7.Note that any function can be expressed as sum of an even and an odd functions.

5 Find the inverse Fourier Transform of F (f) = Setting f = −jp 2π 1 24π 2 f 2 + j 2πf F (f) = = −1 1 = 2 − p2 + p (p + 1)(p − 2) 1 3 p +1 p−2 1 −t f(t) = [e u(t) + e2 tu(−t)] 3 Example 7. The only new aspects of the procedure are as follows: 1. 2.37) The inverse transform of a rational function of f can be found by using a procedure almost exactly like that used for the Laplace transform. the function comes under the category of special Fourier Transforms and caution should be exercised. so that f = −jp/2π and the Fourier transform can be written as a ratio of polynomials in p.34) (7.33) (7.32) (7. we multiply by u(−t).6 Find the Inverse Fourier Transform of F (f) = A f 4 + a4 − 1 3 118 .36) (7. 3.35) (7. The inverse transforms of the terms in the partial fraction expansion that have left half p-plane poles are exactly the same as those in the Laplace transform. Example 7. as those in Laplace transform.31) (7.5 THE INVERSE FOURIER TRANSFORM OF A RATIONAL FUNCTION j 2π (7. The Inverse transform of the terms in partial fraction expansion that have right half p-plane poles are also the same functions. If we use p = j2πf instead of s. If there are poles on the j-axis of the p-plane. but instead of multiplying them by u(t). The inverse transform of rational function p is then found by a partial fraction expansion exactly like that used to obtain inverse Laplace transform of function of s.dF (f) df F(f ) a F[f(at)] = a F[G(t)] = g(−f) F[G(−t)] = g(f) F[f(−t)] = F (−f ) F[fe (t)] = Re[F (f)] F[fe (t)] = jI m[F (f)] F[tf (t)] = 7.

f (t) = ∙ 4 ¸ 16 45o 16 −45o πA √ √ − √ √ + conjugate 2a 3 p + 2πa − j 2πa p − 2πa − j 2πa p+ √ + p+ √ 2πa − j 2πa + conjugate # i √ √ √ πA h −√ 2π at cos( 2πat − 45o )u(t) + e 2π at cos( 2at + 45o )u(−t) e a3 For negative t.where A and a are real positive constants. only the terms with −j in the factors are required. the cosine in the second term can be written as √ √ cos(− 2πa|t| + 45) = cos( 2πa|t| − 45o) since cosine is an even function. " 1 6 −4 5o 32π 3a 3 j 2πa − 2π a 16 −13 5o 32 π3a 3 F (f) = 16π A = so. Then A F (f) = £ −jp ¤4 2π + a4 (2π)4 A = 4 p + (2πa)4 The roots of the denominator are p4 = −(2πa)4 . and 2πa(−1 − j). As already discussed in the treatment of Laplace transform F0 (s) = Fin (s)H(s) 119 (7.38) . then the output f0 (t) will be some function that depends upon the transfer function of linear system as well as upon the input signal. p2 = ±j(2πa)2 √ √ Since the square roots of +j are √ + j)/ 2. 2πa(−1 + j ). as it should be with F (f) being real. we have F (f ) = 16π4 A √ √ √ √ √ √ √ √ (p + 2πa − j 2πa)(p − 2πa − j 2πa)(p + 2πa + j 2πa)(p − 2πa + j 2πa) If this is expanded in the partial expansion.6 CONVOLUTION Suppose that a linear system is excited by an input signal fin (t). ±(1 of √ √ the four roots are 2πa(1 + j). 7. In factored form. Replace f by −jp/(2π). and the square roots√ −j are ±(1 − j)/ 2. Then f (t) may be written as f (t) = h√ i πA −√ 2π a|t| e cos 2πa|t| − 45o (t) 3 a and it is seen to be an even function of t. 2πa(1 − j).

the order in which the functions are chosen is immaterial. convolution for the Fourier transform can be defined. so Y (f ) = F1 (f)F2 (f) Since F1 (f )F2 (f) and F2 (f)F1 (f ) must be equal.and f0(t) = Z ∞ −∞ fin (τ )h(t − τ)dτ (7. Z ∞ Z ∞ f1 (τ)f2 (t − τ )dτ = f2 (τ )f1 (t − τ)dτ −∞ −∞ 120 .1: Let f1 (t) and f2 (t) be Fourier transformable with transforms F 1(f) and F2 (f) respectively The Fourier transform of f(t) = Proof: By definition. this will be same as −∞ and ∞. one in τ alone and one in ξ alone Y (f) = Z ∞ −∞ © f1 (τ)e− j2πdτ dτ ª ½Z ∞ −∞ f2 (ξ)e−j2π f ξ dξ ¾ But this is the product of two Fourier transforms F1 and F2 . then dx = dξ and the limits on the x integration become −∞ − τ and ∞ − τ for the ξ integration. the Fourier transform of y(t) is ¾ Z ∞ ½Z ∞ f1 (τ )f2 (x − τ )dτ e− j2πf xdx Y (f ) = ∞ −∞ Z ∞ −∞ f1 (τ )f2 (t − τ )dτ is F1 (f)F 2(f) If the integrated integral is expressed as a double integral Z ∞Z ∞ Y (f) = f1 (τ )f2 (x − τ )e−j2π f x dxdτ −∞ −∞ If the new variable ξ = x − τ is substituted for x. But for fixed τ .39) In a similar manner. Theorem 7. so Z ∞Z ∞ Y (f) = f1 (τ )f2 (ξ)e− j2π(ξ +τ ) dξdτ −∞ −∞ But the integrand can now be separated into two functions.

1 THE UNIT IMPULSE The unit impulse function δ(t) presents some difficulties.7 Let f1 (t) = e−t u(t) and f2 (t) = e−2 tu(t + 2) convolve f1 and f2 Let y(t) be the result of convolving f1 and f2 .7.40) Example 7.Theorem 7. and y(t) = e 2 e−2tu(t + 2) Z t e y dy = [e 2 e−t − e −2t ] u(t + 2) 7. with transforms F 1(f) and F2 (f) respectively the Fourier transform of f1 (t)f2 (t) is given by. so u(t − y) can be dropped if the upper limit is changed to t. If the function is transformed directly Z ∞ δ(x)e −j2π f x dx = 1 (7. Then.2: If f1 (t) and f2 (t) are Fourier transformable.41) F[δ(t)] = −∞ −2 as in the case of Laplace transform. y(t) = = Z ∞ e− τ e −2t e+2τ u(τ )u(t − τ + 2)dτ Z ∞ eτ u(τ)u(t − τ + 2)dτ = e−2t −∞ −∞ Z−∞ ∞ e− τ u(τ )e− 2(t−τ ) u(t − τ + 2)dτ Let y = τ − 2 or τ = y + 2 y(t) = e2 e− 2t Z ∞ −∞ e y u(y + 2)u(t − y)dy But the integrand is zero for y > t. F [f1 (t)f2(t)] = Z ∞ −∞ F1 (y)F2 (f − y)dy (7.7 SOME SPECIAL FOURIER TRANSFORMS 7. Then Zt y(t) = e2 e −2t ey u(y + 2)dy −∞ The lower limit may now be changed to −2. The difficulty arises when we attempt to inverse transform δ(t) = Z ∞ −∞ e j2 πty dy 121 .

This can only be demonstrated indirectly since the integral does not simply exist in the ordinary sense. If 1 is the Fourier transform of δ(t), then according to Eq. (7.34) δ(f) is the transform 1. It is convenient to define δ(t) as the limiting form of any function of time whose transform approaches unity. There are many such functions and few are listed here. 1. The tall rectangular pulse of height 1 and width . Its transform is known to be (sin π f)/ π f. which certainly approaches unity for any f if is made sufficiently small. 2. The tall triangular pulse of height 1 and width 2 , which has a transform of (sin2 π f)/(π f)2 . This approaches unity for any fixed f so long as is made sufficiently small. 3. The function e−t /2σ2 √ 2πσ has a transform e −2π σ f and can be made to approach unity for any value of f , however large, if σ is made small enough. Then e− t /2σ 2 δ(t) = lim √ σ→0 2πσ is another choice for the definition of the delta function. 4. Since the delta function is real and even any transform that is real and even and that approaches unity in a limit can be used for the transform of δ(t). If we consider the function [U (f − f1 )], which is unity in the region −f1 < f < f1 , so the transform approaches unity for all f as f1 becomes infinite. For any finite f1 , the inverse transform is Z Thus we might define δ(t) = lim sin 2πf1 t πt
f1
2 2 2 2 2

(7.42)

e j2πty dy =

− f1

sin 2πf1 t πt

f1→∞

Although this is a useful definition mathematically, it is somewhat awkward to visualize as a function of time. If the numerator and denominator are multiplied by 2f1 , the relation reads ∙ sin 2πf1 t 2πf1 t ¸

δ(t) = lim 2f1
f 1→∞

(7.43)

Which is known to have value of 2f1 at t = 0. Furthermore, the area under the function can be shown to be unity. Thus the function becomes infinitely high at t = o and it does have the proper area under it. However, the envelope of the function does not go to zero as f1 becomes infinite.
122

5. Another possibility is the transform pictured in Fig. 7.6. Again as f1 becomes infinite, the triangle approaches unity for all f , though it never gets there for any f but zero. We need transform the positive position only; then we take twice the real part, since the transform is real and an even function of f. In the positive region, the function is given by 1−f [u(f ) − u(f − f1 )] f1 and the inverse transform is the same as the transform of 1−t [u(t) − u(t − f1 )] f1 with f replaced by t. The Laplace transform of the triangle is 1 1 e −s f1 − + s f1 s2 f1 s2 with s = j2πf , the Fourier transform is ∙ ¸ £ ¤ 1 1 −1 1 − e−j2π f1f − j2πf f1 (2πf)2 Only the real part is needed, and thus 1/(j2πf) can be dropped. Since 1 − e−j2 πf1 f = 1 − cos(2πf1 f ) + j sin(2πf1 f) the real part of the transform is simply 1 − cos(2πf1 f) 2 sin 2 (πf1 f) = (2πf)2 f1 (2πf)2 f1 Replacing f by −t and doubling the result gives. 4 sin πf1 t (2πt)2 f1 as the inverse transform of the function of Fig. 7.6. 6. There are many new definitions that could be construed for the delta function. For instance e|−f /f1| is an even function of f and approaches unity as f1 becomes infinite. Its inverse transform 2 can be found by noting that the Fourier transform of e−α|t| is α2+4α 2f 2 . If we write −t for f π −|f /f1| . and 1/f1 for α to obtain the inverse transform of e This inverse transform is then
123
2

h i2
1 f1

2 f1

+ 4π 2 t2

Since f1 is to be made large, we might let α = 1/(2πf1 ), and as f1 becomes large α will become small. Then the last expression becomes
α 2(2πα) π = (4π2 α2 + 4π 2 t2 ) α2 + t2

Then α(t) = lim
α→0 2

(7.44) α + t2 This definition agrees well with the initial concept of the impulse, since at t = 0 it has a value 1/πα, which becomes large as α → 0. Also for t 6= 0, the function approaches α/(πt)2 or zero as α → 0. 7.7.2 THE STEP FUNCTION Since the Fourier transform of unity has been defined as δ(f), it should be possible to define the Fourier transform of the step function. Before we begin, the even part of the step function is 1 1 [u(t) + u(−t)] = 2 2 The real part of the Fourier transform of u(t) is δ(f )/2. The transform of u(t) does not exist in the strict sense. The transform of e− αt u(t) does exist; it is 1/(j 2πf + α). This function approaches the step function as α approaches zero, so the transform of step function must be 1/(j 2πf). But this has no real part, yet it has already been established that the real part is δ(f )/2. If what has been done is correct, F[u(t)] = lim
α→0

α π

1 j2πf + α

Multiplying numerator and denominator by (α − j2πf), we have. F[u(t)] = lim
α→0

½

α j 2πf − α2 + (2πf )2 α2 + (2πf)2
1 (j2π f )

¾

The limit of the imaginary part, as indicated before, is dividing numerator and denominator by 4π2
α (4π 2)

, but the real part is, after

Re{F [u(t)]} = lim
α→0

α2 (4π 2 ) + f 2

If we set a =

α 2π

, a → 0 as α → 0, so
124

Re {F [u(t)]} = lim
α→0

a 2π

a + f2
2

But this is exactly one half the expression for δ(f) given in Eq. (7.44) with t replaced by f . The real part of the transform is indeed δ(f )/2 then 1 δ(f) + 2 j 2πf

F[u(t)] = 7.7.3 THE SIGNUM FUNCTION

(7.45)

The function ”signum of t” abbreviated sgn(t) is equal to +1 when t is positive, −1 when t is negative and 0 when t = 0. Thus Sgn(t) = u(t) − u(−t) (7.46)

This function is Fourier transformable in the sense that a constant and a step function are transformable. So that ∙ ¸ 1 δ(−f ) 1 1 δ(f ) + − − = 2 j2πf 2 j2πf jπf

F[sgn(t)] =

(7.47)

Poles on the j axis in the p plane are transformable then, with p = j2πf ∙ ¸ 1 Sgn(t) = 2 p

F

(7.48)

¸ ∙ 1 sgn(t) = F ej2π f0t 2 p − j2πf0 Example 7.8 Find the inverse Fourier transform of 4p4 − 2p3 + 6p2 − 66p − 18 p(p + 1)(p − 2)(p2 + 9) Writing the partial fraction expansion of the above expression 2 1 1−j 1 + − + + conjugate p p + 1 p − 2 p − 3j

(7.49)

1. The left half plane poles have the same inverse transform as the Laplace transform; hence F −1 ∙ ¸ 2 = 2e −t u(t) p+1
125

2. The right half p-plane poles have the same inverse transform as the Laplace transform, but with u(t) replaced by −u(−t). ∙ ¸ −1 F −1 = e2t u(−t) p−1 3. The j axis p-plane poles have the same inverse Laplace transform, but with u(t) replaced by sgn(t)/2. Hence F −1 ∙ ¸ 1 1−j Sgn(t) + + conjugate = [1 + 2cos(3t − 45)] p p − 3j 2

7.7.4 PERIODIC FUNCTION Since the transform of ej2π f0t is understood to be (f − f0), any periodic or almost periodic function that is expressible in an exponential series has a Fourier transform. Thus with
∞ X

f(t) =

Bkej2 πfk
∞ X

(7.50)

k=− ∞

F[f(t)] = F (f) =

k=−∞

Bk δ(f − fk )

(7.51)

126

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7. the functions a) e− 2t u(t) and e3tu(t) b) e− tu(t) and sin 2t u(t) c) et u(−t) and e−t u(t) i2 128 .PROBLEMS 7. and check by transforms. 7.3 Using the transform of triangular pulse in Fig.5 Find the Fourier transforms of a) e− at u(t + b).8 Find the inverse Fourier transforms of [ a ] −f2 i a) h 2π 2 2 a f 2+{ 2π } b) h − j af π f 2+ a { 2π } 2 2 7.33).25f 2 + 0. 7. 7.7.1 Find the Fourier transform of the functions pictured in Fig. a and b are positive b) eat [u(t) − u(t − b)] 7.4 Find the Fourier transforms of a) e− at u(t) − eat u(−t) b) 1 a2 +t2 7.2 Find the Fourier transforms of a) e− a|t| h i πn b) sin(βt) u(t + 2π n ) − u(t − 2 β ) .6 Find the inverse Fourier transform of the following functions. n is an integer β 7.9 Convolve by direct integration.7 Find the inverse Fourier transforms of a) b) 1− π2f 2 f 4+ 1. find the Fourier transform of (sin2 at)/t2 . a) b) c) B f 2+a 2 jB f f 2+b 2 p 2−4 (p+1 )(p+ 2)(p−3) where B and b are real and p = j2πf 7.7(a) and Eq. (7.25 π2 π4 1 (πf +j)2 7.

Hint: It is necessary to show only that several functions have different transforms but the same F (f )|2 .7.13 Given that the Fourier transform of unity is δ(f ). is |F (f )|2 . 7. 7. 7.14 Find the inverse Fourier transforms of a) b) c) p p2+β 2 β p2+β 2 2 p(p+1) 129 . c) Demonstrate that several functions can have the same correlation function. show that regardless of the sign of c.11 Use the theorem on ”complex translation” to demonstrate that 1 2πj Z b+j∞ L[f1 (t)f2 (t)] = b−j∞ F1 (ω)F2 (s − ω)dω provided that b is greater than the abscissa of exponential order of either f1 or f2 . 7. e− bt f(t) has a transform. the Fourier transform of φ(τ). Show that this Fourier transform is F (j2πf + b).12 The auto correlation function φ(τ ) of a function f(t) is defined by φ(τ ) = Z ∞ −∞ f(x)f (x − τ )dx a) Show that φ(τ ) is an even function of τ . find the Fourier transforms of cos(2πf0 t) and sin(2πf0 t). b) Show that φ2 (f ). provided that b > c.10 Let f(t) be Laplace transformable with transform F (s) and having σe = c.

X(f) = 0 for |f| > ω Also. or will simply be the square root of the energy spectrum. The most general signal that satisfies this requirement is the function S(t) = A(t) cos [2πfc t + θ(t)] (8.VIII APPLICATIONS OF THE FOURIER TRANSFORM 8. If an audio or video signal Sm (t) is to be carried by the function of Eq.1) Where A(t) is some arbitrary function of time whose spectrum does not exceed the frequency fc (it is usually small compared to fc) and θ(t) is signal whose maximum derivative does not exceed in magnitude the value 4πfc /3 and whose spectrum does not exceed fc/2.1. let the message be scaled to have a magnitude not exceeding unity.3) (8. In the following subsections. This will simply be the transform of the signal.2) (8.5) . the result is called angle modulation. the Fourier spectrum will be a pair of impulses. above which spectral content is negligible and unnecessary. If θ(t) is a function of Sm (t) and A(t) is a constant. which is plotted graphically in magnitude. the result is called amplitude modulation.1 MODULATION Let S(t) be a signal whose spectrum lies in the vicinity of carrier frequency fc . |x(t)| < 1 or < x2 (t) >≤ 1 The ensemble average then also satisfies X 2 (t) ≤ 1 130 (8.4) (8. In case of sinusoids. a frequency high enough that radiation is economical and practical. 8.CHAPTER . the Fourier spectrum will be used interchangeably with line spectrum or the energy spectrum.1) either A(t) must be some function of Sm (t) or θ(t) must be some function of Sm (t). The two most common examples of angle-modulation are phase modulation (P M ) and frequency modulation (F M ). (8.1 Amplitude Modulation An arbitrary message x(t) can represent the ensemble of all probable messages from a given source. If A(t) is a function of Sm (t) and θ(t) is a constant. Assume that the messages are bandlimited in ω.

7) The spectrum of the modulated signal is shown in Fig. then PT PT = [1 + m2 < x2 (t) >] If the message source is ergodic ¤ A2 £ c PT = 1 + m2 x2 = Pc + 2P sB 2 131 A2 c 2 (8. The message signal and the modulated signal are shown in Fig. the second term averages to zero. The Fourier transform of Eq. 8. Transmission bandwidth BT required for an AM signal is exactly twice that of message bandwidth.6) . 2.9) . If the d − c component of the message is also zero. It is symmetric about the carrier frequency with amplitude as even function and phase as odd function. If m > 1.8) Since fc >> ω.1. which results in carrier phase reversal and envelop distortion. overmodulation takes place. (8.unmodulated carrier (8. The average transmitted power PT 2 2 PT = < Xc (t) >= A2 < [1 + mx(t)] cos2 ωct > c 2 Ac {< 1 + 2mx(t) + m2 < x2 (t) > + < [1 + mx(t)] 2 cos 2ωct >} = 2 (8. The modulated signal is Xc (t) = A [cosωc t + mx(t)cosωc t] = Ac[1 + mx(t)]cosωct where Accosωc t m . (8.7) are as follows: 1.modulation index The modulated amplitude Ac(t) = Ac[1 + mx(t)] fc >> ω and m ≤ 1 When m = 1.6) is Xc(f ) = Ac mAc [δ(f − fc ) + δ(f + fc)] + [x(f − fc ) + x(f + fc )] 2 2 (8.2 The properties of Xc (f) in Eq. 100 percent modulation takes place and amplitude varies from 0 to 2Ac .The envelope of the modulated carrier has the same shape as the message waveform. 8.

.

Power in each side band. 8. c 2 PsB = m2 x2 1 1 A2 c = m2 x2 Pc < P c 4 2 2 This implies that at least 50 percent of the total power resides in the carrier. peak instantaneous power is proportional to 4A2 . The maximum voltage Xcmax = 2Ac and .Where x2 is the ensemble average. find and draw the modulated signal. which is wasted.1 If x(t) = A cos 2πfm t. xc (t) = Ac(1 + mAm cos ωm t) cos ωct mAm Ac [cos(ωc − ωm )t + cos(ωc + ωm )t] = Ac cos ωct + 2 The spectrum of the modulated signal is given in Fig. and carrier is Ac cos ωc t. c Example 8. and carrier power P c is 1 A2 . therefore.3. 133 .

we can always construct the other.2 Double Sideband Suppressed-Carrier Modulation (DSB) The carrier frequency component is independent of message and represents wasted power. we get Xc(t) = x(t)Ac cos ωc t 8. it can be eliminated from the modulated wave without losing any information. (8. (8.4.6) Xc (t) = Accosωct + mx(t)Accosωct Dropping the first term and m in the above equation. c Xc(f ) = Ac [X(f − fc) + X(f + fc )] 2 (8.e. 8. consider.5. However.1. therefore.3 Balanced Modulator for DSB The DSB is obtained by using two AM modulators arranged in a balanced configuration to cancel out the carrier. Assuming that AM modulators are identical save for the reversed sign of one input. BT = 2ω and AM and DSB are quite similar in the frequency domain.1. given the amplitude and phase of one.10) then is PT = 2. Xc(t) = 0 when x(t) = 0 (8. Total elimination of carrier and one sideband from AM spectrum produces SSB for which 134 ∙ ¸ 1 and Ac 1 − x(t) cos ωtc 2 . they are quite different in the time domain. It can be seen that the bandwidth remains unchanged i. 8.PSB = The peak power is proportional to A2 then. Eq. the outputs are ¸ ∙ 1 Ac 1 + x(t) cos ωct 2 Subtracting the two components. Therefore. transmitting both bands is a waste of bandwidth. we get Xc (t) = x(t)Accosωct so that.8. The arrangement is shown in Fig.10) The average transmitted power for Eq.11) X 2 A2 c 2 The spectrum of the modulated signal is shown in Fig. 8.1.4 Single Sideband Modulation The upper and lower side bands of AM and DSB are uniquely related by symmetry.

.

.

we get 1 − [cos 2π(106 − 220)t + cos 2π(106 − 440)t] 2 Multiply Eq. The resulting DSB −SC is filtered so that only the frequencies less than 1M C are retained.110t + cos 2π. In normal AM detection. the modulating signal is recovered by applying Xc(t) to a half wave rectifier. 137 . What will the output be now when the signal is multiplied by cos 2π(106 − 110)t and the frequencies above the audio range are filtered out. 8.9. A scheme of demodulation is shown in Fig.12) by cos 2π(106 − 110)t.1.8 Example 8. we get Xc (t) = 1 [cos 2π. 8. 8.7. The diode in Fig.9. This will indicate what happens when music is received by a receiver with drifting oscillator.BT = ω and PT = PsB = X2 Ac 4 The arrangement for obtaining SSB is shown in Fig. so that frequencies below 1MC are retained. 8.6 and the spectrum is shown in Fig.2 A modulating signal x(t) = cos 440πt + cos 880πt is multiplied by a carrier fc = cos 2x106 πt. The output is then filtered to provide the desired modulating signal.330t] 4 1 = [cos 220πt + cos 660πt] 4 (8. This is inverse of modulation and requires time varying or nonlinear devices. A balanced SSB modulator is shown in Fig.12) This indicates the deterioration in SSB because of corruption of carrier by 110Hz.5 Demodulation or Detection The process of separating a modulating signal from a modulated carrier is called detection. 8. can be treated as a piece wise linear device where switching takes place at carrier frequency f . (8. 8. The SSB is widely used for transoceanic radio telephone circuits and wire communication. Solution: The DSB is given by the equation (cos 440πt + cos 880πt) cos 2πx106 t = 1 1 [cos 2π(106 + 220)t − cos 2π(106 − 220)t] + [cos 2π(106 + 440)t − cos 2π(106 − 440)t] 2 2 Filtering.

contains a component proportional to x(t) plus higher frequency terms.Thus. block to remove the bias of unmodulated carrier component. Now let the instantaneous frequency vary linearly with x(t) ωi = ωc + k 2x(t) ωidt = ωct + θ0 + k2 θ(t) = Z Z x(t)dt (8.17) Where k1 is a constant of the system. 8. It can be shown that the output of Fig. In this type of modulation. the carrier must be supplied at the receiver before detection can take place. The time constant is much larger than 1/fc and smaller than 1/ω. 8. The capacitor serves to filter out the higher frequency terms. In particular. the frequency of the carrier is caused to vary according to the modulating signal x(t). Thus the frequency of the carrier is ωc + kx(t). Both phase and frequency modulation are special cases of angle modulation. Strictly speaking. we call the resulting form of modulation as angle modulation. 138 . S(t) is the switching function. R2 C2 acts as a d.13) (8. if θ(t) = ωc t + θ0 + k1 x(t) (8.8. we can talk of only sine(cosine) waves for understanding this type of modulation. The sum of the signal and locally generated carrier could be rectified to select the components corresponding to the desired modulating signal. It is more common in practice to use the carrier to shift the SSB signal to required audio band by using a frequency converter.18) This gives rise to F M system.15) When θ(t) does not vary linearly. The problem of providing a carrier at the receiver of exactly the right frequency has been a block in the wide spread use of SSB.14) The filtering is carried out by the low pass filter with time constant R1 C1 . In case of SSB detection.16) If θ(t) is now made to vary in some manner with a modulating signal f (t).1. Xc(t) = k[1 + mx(t)]cosωct and 0 Xc(t) = k[1 + mx(t)]cosωct. Thus fc (t) = cosθ(t) = cos(ωct + θ0 ) (8. Here the phase of carrier varies linearly with modulating signal.S(t) (8. If the angle varies linearly with time. we can obviate this difficulty by defining instantaneous radian frequency ωi to be the derivative of the angle as function of time ωi = dθ(t) dt (8. the frequency can be expressed as the derivative of the angle.c.6 Frequency Modulation.

Thus the bandwidth of F M depends on β. new frequencies are generated by the modulating process. so that Xc (t) = cos(ωct + β sin ωm t). In this case β is usually smaller than 0. ∆f = ω/2π gives maximum frequency of deviation called frequency deviation.2. The average power associated with the F M carrier is independent of modulating signal and is the same as average power of the modulating carrier.19) ∆ω << ω β is called modulation index and represents maximum phase shift of the carrier. The average power over a cycle for 1 ohm is 1 T Where Z T 2 Xc (t)dt = 0 1 T Z T cos2 (ωct + β sin ωm t)dt (8. The equations for narrowband F M appear in the form of product modulator of AM and give rise to sideband frequencies equally displaced about the carrier. β= ∆ω ∆f = ωm fm (8.1. consider a sinusoidal modulating signal at fm X(t) = a cos ωm t The instantaneous radian frequency ωi ωi = ωc + ∆ω cos m t where ∆ω is a constant depending on the amplitude 0 a0 of the modulating signal and circuit converting variations in signal amplitude to corresponding variations in carrier frequency.7 Narrowband F M In this case β << π/2. So that 139 . θ0 may be taken as zero by referring to an appropriate phase reference. As the simplest example. The phase variation θ(t) for this special case Z ∆ω sin ωm t + θ0 θ(t) = ωidt = ωct + ωm Here. This result is true for c general form of signals.20) 0 1 fm Z 1 T 1 + cos(2ωct + 2β sin ωm t dt = T 0 2 1 watt = 2 If the amplitude of the carrier is Ac. Thus ωi varies around ωc at the rate of ωm and with maximum deviation ∆ω. T= 8. the average power is 1/2 A2 .Since F M is a nonlinear process.

For β << π/2 Xc(t) = cos(ωct + β sin ωm t) = cos ωc t cos(β sin ωm t) − sin ωct sin(β sin ωm t) (8.21) cos(β sin ωm t) = 1. carrier and sidebands are in phase.fm . The bandwidth required to pass this signal becomes correspondingly large.1. Whereas in AM.10. and sin(β sin ωm t) = β sin ωm t Eq. 8. carrier and sideband terms are in phase quadrature. Consider β >> π/2 Xc (t) = cos ωct cos(β sin ωm t) − sin ωc t sin(β sin ωm t) 140 . Xc (t) = cos ωct − β sin ωm t sin ωct β = cos ωct − [cos(ωc − ωm )t − cos(ωc + ωm )t] 2 ¸ ∙ β β jω c t = Re e (1 − e −jω mt ejω mt) narrow band F M 2 2 Xc (t) = cos ωct + mx(t) cos ωct (AM ) 8. so that π 2 Xc (t) = cos ωc t − k2 g(t) sin ωct |k2 g(t)| << (8. For general signal Taking θ = 0 and R ωi = ωc + k2 x(t) Z Z θ(t) = ωi dt = ωc t + θ0 + k2 x(t)dt x(t)dt = g(t) Xc(t) = cos [ωc t + k2 g(t)] If k2 and the amplitude of G(t) are small.21) can now be written as Xc(t) = cos ωc} − β sin ωm t sin ωc t | {z t | {z } Carr ier Sideband f requencie s Thus the BW of narrowband F M is 2. (8. This is demonstrated by Fig.8 Wideband FM The advantage of noise and interference reduction of F M over AM becomes significant for β >> π/2. where fm is the highest frequency component of either g(t) or its derivative x(t).22) The bandwidth is again 2fm . In F M .

.

β is proportional to the amplitude of the modulating signal. whereas. increase in bandwidth and sidebands is accompanied by decrease in power in the carrier and hence the amplitude of the carrier is decreasing.23) is shown in Fig. For β < 0. Since the average power is constant. Xc(t) = cos(ωct + βsinωm t) = cos ωc t cos(β sin ωm t) − sin(β sin ωm t) sin ωc t Both cos(β sin ωm t) and sin(β sin ωm t) are periodic functions of ωm and each may be expanded in Fourier series of period 2π/ω. we get the additional term sin 2 ωm t cos ωct in Xc(t) so that Xc(t) = (1 − + β2 β ) cos ωct − [cos(ωm − ωc)t − cos(ωc + ωm )t] 4 2 (8.50.23) β2 [cos(ωc + 2ωm )t + cos(ωc − 2ωm )t] 8 The amplitude spectrum of Eq. sideband for cos ωct will be in phase. we require more terms in power series expansion of both cos(β sin ωm t) and sin(β sin ωm t) and bandwidth begins to increase with β. Each term will have terms in ωm and all its harmonics and each harmonic multiplied by cosωc t or sinωc t gives rise to two side bands symmetrically situated about ωc .11.Since β is significant cos(β sin ωm t) ' 1 − If we assume β << √ β2 sin2 ωm t 2 β 2 << 6 6 and retain just the first two terms. Consider. For a fixed modulating frequency. component plus a small component at twice the fundamental frequency. (8. Note that the carrier term has decreased somewhat with increasing β. 8. The sidebands for sin ωc t will be quadrature apart. As β increases further. Let us consider cos(β sin ωm t) 1.c. the curve can be represented by d. 142 .

8.c.2. so that they give rise to odd order sidebands about the carrier.10 Pulse Modulation Consider a periodic function δp(t) consisting of impulses occuring every Ts sec and having an area of Ts units.24) gives cosine terms and the imaginary part gives sine terms. (8. This function is shown in Fig. Consider periodic complex exponential V (t) = ejβ sin ω m t. 8.14. − T T <t< 2 2 (8. This device is called a frequency discriminator.24) The real part of Eq. 1 T Z T /2 Cn = 2π T ej(β sin ωmt− ωn t) dt −T /2 (8.9 Frequency Demodulation The demodulation process must provide an output voltage (current ) whose amplitude is linearly proportional to the frequency of the input F M signal. the carrier term decreases with increasing β and sidebands increase. component with some ripple superimposed. For β > π/2. So. We can decrease the carrier power (wasted power) considerably by increasing β. The Fourier series of δp(t) can be obtained by finding the Laplace transform of one period and letting s = j2πk/Ts .1.26) The integral in Eq. Then C k = Ts and the exponential series is 143 .26) can be evaluated only as an infinite series and is called Bessel function of the first kind and is denoted by Jn (β) = 1 2π Z π ej(β sin x−nx) dx −π A circuit for direct F M modulation is given in Fig.25) ωm = 2πn ωn = nωm = T Z 1 π j(β sin x− nx) = e dx 2 −π x = ωm t (8. If we multiply this by cosωct.12. A circuit of a discriminator is shown in Fig. the function takes on negative values. as β increases positive and negative excursions become more rapid. 8. at β = π/2 transition from more or less slowly varying periodic time function with most of spectral energy in its carrier to a rapidly varying function with the spectral energy spread over a wide range of frequencies. the frequency components are all old integeral multipliers of ωm . 8. For β > π/2. (8. Consider sin(β sin ωm t). 8.13.1. 3. the function remains positive and appears as a d. But Laplace transform of one period is simply Ts .

.

31) k=−∞ The Fourier transform of Eq. In general. . where fm is the maximum frequency component of Sm (f). and 2fm wide will produce the original AM − SC .27) ∞ X s0 (t) = sm (t)ej2π kt/Ts (8. 2/T s. 3/T s etc. ... (8.” Note that although a low-pass filter will restore the original signal.31) is s0 (f ) = ∞ X k=−∞ s m (f − k ) Ts (8. 8. so δp(t) = Σ∞ Ts δ(t − rTs ) r =−∞ s0 (t) = ∞ X (8. 2/T s. the product will be a series of impulses seperated by TS sec.32) Then.28) (8. Typical spectrum of Sm (f) and S0 (t) = Sm (T )δP (t) are shown in Fig. Since impulses are impossible to produce. the spectrum of the sampled signal is the same as Sm (f ) translated to the right and left 1/T s. pulses of suitable width can be used for this purpose. The result is only a sampling Sm (t) at the sampling instants. Sm (t) can be sampled at any rate greater than or equal to 2fm and original signal can be recovered from its samples by filtering. a bandpass filter centered at f = 1/T s. In addition to Sm (f ) itself when k = 0. if the pulse width is ∆t. the Fourier transform of 145 . s0 (t) = sm (t)δP (t) but. but whose areas are now Ts times the height of amplitude of Sm (t) evaluated at the time of concurrence of each impulse.15. Then amplitude modulation can be produced by pulse modulation and filtering as well as by using nonlinear devices.29) (8. Mathematically. . It is essential and follows from the sampling theorem which states ”if the maximum frequency component in Sm (t) is fm .δp(t) = = ∞ 1 X j2 πkt/T s e T k= −∞ ∞ X ej2π kt /Ts (8.30) r= −∞ Ts sm (rT )δ(t − rT ) on using Eq. Note that 1/Ts ≥ 2fm .27) k=−∞ If a function Sm (t) is multiplied by δp(t). (8. it can be shown that.

The plate modulator has the added advantage that nearly 100 percent modulation can be achieved with relatively little distortion. The triode is biased well below cutoff. where fm is maximum frequency contained in Sm (f ). the amount of current produced will be proportional to the plate voltage at the moment of conduction. it can be considered an impulse as far as the band of frequencies near fc is concerned. 4 The integrand is very nearly equal to p(t) for all t within the range of integration. Then P (f ) = = Z ∆t/2 p(t)e−j2πf t dt −∆t/2 ∆t/2 (8. so the carrier frequency and the sampling frequency are the same. so when no carrier signal is applied to its grid. The current pulses produced then pass through the bandpass filter represented by the L − C circuit. therefore its transform is real and the second integral vanishes. it is easier to prove: Let p(t) be a pulse of length ∆t and p(t) be an even function of t. With 360 deg to a cycle.16. so even at the upper limit. Even if the condition of symmetry is not met. the pulse should not be larger than 28. for all intents and puposes. there is still a pulse length which the spectrum is very nearly a constant upto some maximum frequency. 8. provided that the pulse is symmetric about its centre. if the objective of sampling is to recover Sm (t) by ultimately using a low pass filter. very nearly P (f ) = for f < 1 4π ∆t Z ∆t/2 p(t)dt = area of p(t) − ∆t/2 Now. The pulse has to be about 8 percent of the carrier period. 146 . this is an ordinary AM signal. cos 2πf t = cos 1 = cos 14. If the sampling is to be used to produce AM . The modulating signal is placed in series with the d-c supply of the circuit. the triode does not conduct at all. wide.3 = 0. The plate modulator operates on this principle and is shown in Fig.969. The L − C resonant circuit provides a low impedance path for the modulating signal and a high impedance to the carrier. The carrier is applied to the control grid with an amplitude that insures that the tube will conduct only when the crest of the carrier is reached. If it conducts.33) Z ∆t/2 Z −∆t/2 p(t) cos(2πf t)dt − j p(t) sin(2πf t)dt − ∆t/2 Since p(t) is even. and the output voltage will be proportional to [Ebb + Sm (t)] cos 2πfct If the maximum amplitude of Sm (t) never exceeds Ebb . it is only necessary that ∆t < 1/(4πfm ).6 deg. So. Then Z ∆t/2 P (f) = p(t) cos 2πftdt −∆t/2 1 But for f < 4π1∆t . For symmetric pulses. so the pulse will have an area under it proportional to Ebb + sm(t).the pulse is equal to the area under the pulse for all frequencies less than 1/(4π∆t). the first repetition of the spectrum is the one that is used. so the plate voltage on the triode is very nearly Ebb + Sm (t). The resonant circuit is also the filter that removes all but those frequencies near the carrier frequency. 2πf < 2∆t . Further-more.

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To use this bandwidth efficiently.Another practical application of pulse modulation is time multiplexing. Observe that the spectrum of the pulses that carry the sampled information must be transmitted by this channel without distortion. whose magnitude is constant over this frequency range. but rather to establish the design criteria for them. Since 2. A physically realizable filter is one whose impulse response is necessarily zero for t less than zero.36) However. If it is possible to send a message by sending only the samples. For example.5KHZ low pass filter. 148 . Its transform H(f ) can be expressed as H(f) = R(f) + jI(f) and Where R(f) and I(f) are the real and imaginary parts respectively. then X0 (f) should be proportional to X(f). and both signals are thus recovered simultaneously. if the device is to produce the signal without distortion. FILTERS Filters are an essential part in the design of linear systems and are used to modify the signal or eliminate the unwanted frequency band. Indeed any communication system involves filters. (8. These might be two voice signal on a telephone line. put both the signals through 2. the criterion for distrortionless filtering is a phase function that is linear with frequency. so that those belonging to one message are alternated with those of the other message. It will tell us what is possible and what is not possible.35) (8.17(b) This pulse (note that its fundamental frequency is 10K HZ) and the pulses at the other end of the line are separated out by some type synchronized switches device or commutator. it is necessary to send more than two message simultaneously. we have used filters in communication systems. In this case magnitude of the transfer function has to be considered.5KHZ is the highest frequency needed or used in voice transmission. 8.1. Then the transform of the output signal will be X0 (f ) = AX (f)ejθ (f ) (8. Let the transform of this signal be X (f) and the magnitude of the transfer function be A. it is not easy to specify criteria for physical realizability. is there not some use that can be made of the time in between the samples? Consider two signals Sm1 (t) and Sm2 (t). But θ(f ) = −j2πft0 would not be X0 (f ) = AX(f)e− j2π f t0 (8. which is the input signal changed in amplitude by A and delayed t0 sec. 8. in frequency domain. Eq. if h(t) is the response of a realizable filter.5KHZ low-pass filter and then sample each at 5KHZ rate. than h(t) = 0 for t < 0.17(a) and the alternating sample pulses appear in Fig. but can be any thing at all other frequencies. 8. Suppose a band limited siganl in |f | < fc is put through a filter.37) H(f ) = |H(f )|e jθ (f ) (8. Assuming that time delay of t0 seconds is not objectionable. However.2. For example. so the channel bandwidth must be determined not by signals being modulated but by the pulses. a bandpass filter should have constant magnitude over the passband and zero magnitude outside the band. 8. Now stagger the sampling pulses.34) has an inverse transform. The Fourier transform is not used to design the filter. This means that θ(f) = 0. Two such signals appear in Fig. and explain some characteristics of practical filters.35) is used in the filter theory. then each is passed through a 2. In Sec.

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39) where fc is the cut off frequency and t0 is the delay time. If the output is to look anything like a pulse. If the integral converges. However. The magnitude of the transfer function is constant in the passband and zero otherwise.38) converges that is.41) This function resembles a step function and is shown in Fig. The criterion that the amplitude of the transfer function must meet to insure that the impulse response will be zero for negative t is called the Paley-Weiner condition.39) is h(t) = A sin [2πfc(t − t0 )] π t − t0 (8.19. the ideal filter can be approached arbitrarily closely if one is willing to increase the delay time t0 . there will always be an overshoot and an undershoot at the discontinuity. This peculiarity is known as Gibbs phenomenon. This states that the |H(f)| may be the magnitude of the Fourier transform of a function which is zero for t less than some time t if and only if the integral Z ∞ −∞ ln|H (f)| df 1 + f2 (8. however that no matter how large fc is made. If fc is made large. there exists a phase function. This leads to the rule of thumb: the bandwidth of a filter must be at least the reciprocal of the pulse length if the pulse is not to be seriously altered in amplitude.40) The function is shown in Fig. 8. The phase function. If the delay time t0 is large 150 .2. Notice. then the step function rise time should not exceed one-half the pulse length. then output can be viewed as positive step followed by a negative step. As with all ideals.18 and it starts wiggling. Then the rise time of the output is π/a = 1/(2fc). Observe that rise time of the step response is very nearly equal to the reciprocal of the slope of {2fc π(t − t0 )} at t = t0 . If the input were a rectangle pulse instead of a step function. 2. 8. if it is less than infinity. Even though the filter above is not realizable. not necessarily linear. so that its inverse transform is zero for negative t. that can be associated with |H(f)|. the frequency of wiggles is large. The inverse transform of Eq. before t = 0 and hence is not realizable. 8.The ideal bandpass filter then has the following properties: 1.1 The Ideal Low-Pass Filter The transfer function of the ideal low pass filter is H(f ) = A [u(f + fc) − u(f − fc )] e−j2π f t0 (8. The unit function response of this filter is A π Z t −∞ sin{2πfc(t − t0 )} dt = Aξ[2πfc(t − t0 )] t − t0 (8. (8. an ideal filter is unattainable. it will be shown that realizable filters demonstrate this overshoot if the magnitude of the transfer function falls rapidly in the vicinity of fc . θ(f) must be linear function of frequency in the passband with negative slope of −2πt0 .

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the result will not have linear phase.21. If the new variable x = f − y is used. with dx = dy and the limit changed to f + fc and f − fc Z f − fc A −j2π f t0 − sin 2πt0 x e dx π x f + fc Z f + fc A sin 2πt0 x = e −j2π f t0 dx π x f − fc Hai(f ) = A{ξ{2πt0 (f + fc)} − x{(2πt0(f − fc ))}}e−j2π f t0 Haif = (8. they may be brought outside the integral sign.sin{2πfc(t− t0 )} t − t0 will be small for negative t. If Hai(f ) is the transfer function of this realizable approximation to the ideal filter. we should be able to approximate the ideal filter by making the impulse response of a filter. 8. The transform of this function can be approached in two ways. Symmetry can be achieved however by chopping the response of ideal filter for t > 2t0 as well. t≥0 π t − t0 Unfortunately this function is not symmetrical about the time t = t0 .20. The impulse response above can be viewed as the original impulse response multiplied by {u(t) − u(t − 2t0 )}.42) (8. the phase function has discontinuities in it of magnitude π every time a change in sign occurs. The transform of the resulting function is sin 2πf t0 −j2 πf t0 e πf and this can be convolved with the transform of the ideal filter to find the transfer function of almost ideal filter. A sin 2π{fc(t − t0 )} . Therefore. 152 . then Z fc Hai f = = Z − fc fc Ae−j2π yt0e −j2π (f −y)t0 sin 2π(f − y)t0 dy π(f − y) A sin 2π(f − y)t0 −j2π f t0 e dy π(f − y) − fc Since the exponential and the constant under the integral sign are independent of y. so when this is transformed. (8. Notice that since the function becomes negative for alternate intervals below −fc and above fc . 8. The resulting response is shown in Fig.43) The magnitude and phase of Eq.This in no way detracts from the linearity requirement of the phase.43) are plotted as a function of frequency in Fig.

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then to convert it to low pass filter with cutoff frequency fc and impedance level R0 we simply Multiply all resistances by R0 Multiply all inductances by R0 /(2πfc ) 154 . This appears in Fig. The larger this slope. Suppose we have been given a low pass filter constructed of R. 8. it is necessary only to design the equivalent low pass filter with a cut off frequency of one rad/sec and one Ohm impedance level. so there was little concern for linear phase functions. where f0 is the centre frequency of the filter. 8. A filter will be function of the slope of the frequency response at the cutoff frequency. 8. the resulting transfer function will be physically realizable.2. Therefore.2. then. but dissatisfied with the impedance level. and we know its circuit diagram. if we subtract the transfer function for the low pass filter from the function given. Thus Hn (f ) = Ae −j2π f t0 {1 − ξ{2πt0 (f + fc) − ξ{2πt0 (f − fc)}} will have an impulse response equal to an impulse value A at t = t0 . If we are satisfied with the cutoff frequency. and C. this result can be made to apply approximately to any signal applied to a band filter. What would happen if all the inductances and capacitances were removed and replaced by inductances and capacitances half as large?. Further. 8. 2.2.4 Practical Filters The first filter designs were used in audio work. then the impedance level can be raised by a factor 0 b0 . the cutoff frequency of the low pass filter being less than the high pass filter. the larger will be the delay time. we design a low pass filter with cutoff frequency 1/(2π)Hz and a one ohm impedance level.2 The High-Pass Filter A filter with transfer function Ae −j2π f t0 for all frequencies is physically realizable. A bandpass filter can be visualised from a low pass filter by simple translation to the left and the right of f0 Hz. 8. If. This means that impulse response of the filter is the same as that of low pass filter multiplied by cos 2πf t0 . The filter will have the same characteristics.22(a). 8. if L and C terms are reduced in size by a factor 0 a0 then the transfer function of the network will have the same amplitude and phase variation. This means that R and L terms are multiplied by 0 b0 but the C terms are divided by 0 b0. L. The magnitude and phase functions for this filter are shown in Fig. Its integral will be step response which will be upside down version of the low pass step response. but with the frequency axis multiplied by the constant 0 0 a . Therefore.We can now draw two conclusions from the result: 1. L and C elements. The response of such filter will be accompanied by about 9 percent overshoot and undershoot at points where the input is discontinuity. We proceed as follows. the price for good low pass filter would appear as a delay in response. minus the impulse response of the low pass filter with the same cutoff frequency. but the bandwidth will be doubled.22(b). If a bandpass or high pass filter is to be designed using lumped elements R.23. In fact. but with a positive discontinuity of A at t = t0 as shown in Fig.3 The Bandpass Filter The bandpass filter may be thought of as the result of subtracting from a constant both a high pass and a low pass filter.

Divide all capacitances by 2πfc R0 If we wish to design a bandpass filter with impedance level R0 and bandpass fc then it is necessary to design the corresponding low pass filter with the same impedance level and cutoff frequency fc and then place (1) in series with every inductance a capacitance that is in series resonant with it at the desired centre frequency f0 and (2) in parallel with every capacitance of the low pass filter an inductance that is parallel resonant with it at the center frequency f0 . The transfer function has thus shrunk in size. then. the new cutoff frequencies are related to each other by fH − fL = fc and fH fL = f02 so that the bandwidth is exactly fc . if it is required to design a high pass filter with cutoff frequency fc and impedence level R. The phase angle will change sign however. Since high pass and bandpass filters can be obtained from low pass filters. but the centre frequency is at the geometric mean of the upper and lower cutoff frequencies. we will consider only the design of low pass filters and compare with the ideal filter. the upper cutoff frequency of the bandpass filter will be about f0 + fc/2 and the lower cutoff frequency near f0 − fc /2. Finally. This means that the impedance of an inductance is replaced by a series resonant circuit. It can be shown in similar way that placing an L in parallel with each capacitance is equivalent to replacing the admittances j2πf C by j2πC (f − f02 /f) where inductances in each case are related to C by 1 L= 2 2 4π f0 C in order that each pair will be resonant at the frequency f0 . but the resulting bandwidth is the same. Actually. This amounts to replacing jωL = j2πf L by j2πL(f − f02 )). On either side of the frequency fc the variation of the impedance of the new elements with frequency will be just the opposite of those they replaced and so opposite transfer function will be obtained. 155 . then we need only design a low pass filter with the same level and cutoff frequency and then replace all C terms by L terms that are resonant with the C terms at the cutoff frequency fc and replace all L terms by C terms that are resonant with L terms at cutoff frequency. however. the function (f − f0 )(f + f0 )/f behaves like 2(f − f0 ) and so if the cutoff frequency of the original lowpass filter is fc. For frequencies near f0 . Both of these operations can be expressed mathematically by saying that the frequency f is replaced by f− f02 (f − f0 )(f + f0 ) = f f This is not quite equivalent to translation to the left or right for bandpass filters. then jωL = j2πf L is replaced by j2πL − j/2πfc where C is related to L by C= 1 4π2 f02 L if it is resonant with it at f0 . It is as though the frequency f were replaced by fc2 /f.

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47) and keep the left half s-plane poles. . the roots of HB 3 (s) are 157 . 2. depending on whether n is even or odd. and it can be shown that those lying in the left half plane are ½ π(1 + 2k) 2n ¾ ½ π(1 + 2k) 2n ¾ Sk ≡ sin for k = 0. . .47) Now if HBn (s) is the transfer function of a realizable filter. then 1 (s − s0 )(s − s1 ) . n − 1 + j cos (8.49) Find the transfer function and step response of a third order butterworth filter.3 (8.48) If these roots are put in the appropriate factors.8. Eq. and throw the others away.45) can now be written as HBn (jω)HB n (−jω) = 1 ω2n (8. It is then necessary only to factor the denominator of Eq. then all its poles must be in the left half plane. (8.46) If we now go backwards and put s = j ω or ω = −js. (8. 1. Thus the roots lie on the unit circle in the s-plane.2. and this is called its cutoff frequency. . Solution: Since n = 3. It is also well known that 2 ∗ |HBn (jω)| = HBn (jω)HB n (jω) = 1 1 + ω2 n (8. (8. . Then HBn (−s) must have all its poles in the right half plane. .44) √ It is seen that the magnitude of the transfer function is 1/ 2 at ω = 1 rad per sec.46) will read HBn (s)HBn (−s) = 1 1 = 1 + (−js)2 n 1 + (−1)n s2n (8.45) But the conjugate of HB n (jω) is HBn (−jω). The roots of the denominator that are in the left half s-plane are the (2n) the roots of −1 or +1. Eq. (s − sn ) HBn (s) = Example 8.5 Butterworth Filters The low pass Butterworth filter of order n has a transfer function with a magnitude of 1 (1 + ω2n )1/2 |HB n (jω)| = (8.

.24 along with the response of the ideal filter. − + s s +1 s+ 1 − j 3 2 2 plus the conjugate of the last term. 8.. This makes the response √ 2 3t + 900 ) 1 − e−t + √ e −t/2 cos( 2 3 √ 2 3 t = 1 − e− t + √ e−t/2 sin 2 3 This response is shown in Fig.s 0 = − sin = s1 = = s2 = = Then the denominator polynomial is " and HB3 (s) = π π + j cos 6√ 6 1 3 − +j 2 2 3π 3π + j cos − sin 6 6 −1 5π 5π + j cos − sin 6 √ 6 3 − 1/2 − j 2 √ #" √ # 1 1 3 3 s+ −j s+ +j (s + 1) = (s2 + s + 1)(s + 1) 2 2 2 2 1 s 3 + 2s2 + 2s + 1 The unit step response will be the inverse Laplace transform of HB 3 (s)/s. that is 1 h √ ih √ i 1 s(s + 1) s + 2 − j 23 s + 1 + j 23 2 = 1 √ 6 90o 1 1 3 √ + . The low frequency group delay of the Butterworth filter can be calculated by noting that s = jω. the transfer function is HB3 (jω) = n 1 1 2 (1 + jω) + j(ω − √ 3 2 on o √ 1 ) + j(ω + 23 ) 2 for which angle is 158 .

but there is no undershoot. By using circuit analysis.25 is a Butterworth third order filter. The overshoot of the Butterworth filter is 8 percent. Solution: 1. this is a factor of 23 per octave or 18db per octave. it can be shown that 1 + s/2 + 3s [1 + 4s(1 + s/2)/3] /2 Is (s) = V0 (s) 1 1 V0 (s) = 3 Is (s) s + 2s 2 + 2s + 1 which is the HB N (s) The magnitude and phase functions are |HB3 (j ω)| = √ and θ(f) = − tan −1 ω + tan− 1 (ω − Since HB3 (s) = 1 1 + ω6 √ √ 3 3 + tan−1 2(ω + ) 2 2 1 (s + 1)(s2 + s + 1) 159 . this is 2 sec. 2.θ(f) = − tan− 1 ω − tan−1 2(ω − √ √ 3 3 ) − tan −1 2(ω + ) 2 2 The negative derivative of this with respect to ω will yield the group delay tg . and so the response of the ideal filter is drawn with this delay.4 1. Since the transfer function falls off like 1/f 3 for large frequencies. Plot the magnitude and phase of its transfer function. Use circuit of (2) to design a bandpass filter with a bandwidth of 15KHz but centered at 100 KHz with a 10k impedence level. 8. 3. That is tg = − 1 dθ(f ) 2 2 = + h h i + dω 1 + ω2 1 + 4 ω − √ 3 2 1 + 4 ω + 2 i2 √ 3 2 At ω = 0. 4. Use circuit of (2) to design a high pass filter with a cutoff frequency of 15KHz and an impedence level of 10k. Show that the transfer function V0 (s)/Is(s) for the circuit shown in Fig. Use the filter given to design a low pass filter with an impedance level of 10K and cutoff frequency of 15KH z. Example 8.

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8.27. 3. The The Chebyshev polynomials are defined by Cn (x) = cos(n cos−1 x) 161 (8. The circuit diagram is shown in Fig.29. Since the level is to be raised to 10k = 104 multiply R and L by 104 .15. 8. 8.6 Chebyshev Filters −2 2 (9π) h. The 1 frequency level is to go to 15. put an inductance of 1/3 of last value in parallel with the 10−2 /(2π)µf capacitance and obtain the circuit in Fig. put a C such that 900 1 = µf 4π2 f02 L 16π 1 3 h = 4π 2 f02 200π In Parallel with the 10−2 /(6π)µf capacitance put an L= Finally. and the µf capacitance becomes L = 2 3π h.2.103 H z from 2π R = 1 to R = 10 L= 4 4. or C = µf . For a bandpass filter at f0 = 105 . For the highpass filter with cutoff at 15kH z replace L and C terms by elements resonant 4 with them at 15kHz. 8.2π. L = π h is replaced by C= 1 4π 2 f02 L 10−2 = µf 4π 10 −2 (6π ) C = 10 π) µf is replaced by L = (2 circuit is shown is in Fig.10 6π 6π 10−2 2π goes to three times the latter.28.104 4 to L = h = 3 3. 4.26 θ(f) = − tan −1 ω + tan−1 2. divide C by 104 .103 9π C= and c = 3 2 1/2 10−2 1 10−8 to C = 4 = µf = 3 2 10 .2π.50) . 8.Then HB3 (jω) = and θ(f ) could be written as 1 (1 + jω)(1 − ω2 + jω) ω (1 − ω2 ) The magnitude and phase functions are shown in Fig. in series with C= 4 9π h.15.

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A device where = 1 leads to a 3 db variation in the transfer function in the passband.54) (8.55) (8.51) C 2 (x) = 2x(x) − 1 = 2x2 − 1 C3 (x) = 2x(2x2 − 1) − x = 4x3 − 3x (8.52) (8. (8. Letting n = 0 in Eq. The nth-order Chebyshev filter has the general form |Hcn (jω)| = p 1 2 1 + 2C n (ω) (8.53) (8.50) C0 (x) = cos(0) = 1 Letting n = 1 gives C1 (x) = cos(cos− 1 x) = x Now C2 can be found by Eq.It can be shown that these polynomials satisfy the recurrence formula Cn = 2xCn−1 (x) − Cn−2 (x) and so if the first two can be obtained then. . the others also can.56) Where is commonly chosen to be less than or equal to one. then the roots of the resulting denominator that lie in the left half s-plane can be shown to be at ½ ¾ ½ ¾ a+1 π(1 + 2k) π(1 + 2k) (a − 1)/a sin + j a cos 2 2n 2 2n Sk = (8. with ω set equal to −js. . . As is in the case of Butterworth filters the magnitude squared is formed.57) where k = 0. 164 . The polynomials are odd if n is odd and even if n is even. .51) and These polynomials are useful because in the interval −1 ≤ x ≤ 1 the polynomials oscillate back and forth from +1 to −1. n − 1 and a= (∙ 1+ 1 2 ¸ 1/2 + 1 )1/n (8.58) These poles lie on an ellipse whose semi-major axis lies on the j axis and whose length is (a + 1/a)/2 and whose semi minor axis lies on the real axis is (a − 1/a)/2 in length. (8. 1. and this is usually considered large. and are always equal to +1 at x = 1 and ±1 at x = −1.

then C3 (ω)/4 is 4ω6 − 6ω4 + 9ω 2 /4. 8.31 shows the step response of the ideal filter having 2. Then from Eq.58) 1 (1 + 9ω 2 /4 − 6ω4 + 4ω6 ) 1/2 √ √ 5+1 a = { 5 + 2}1/3 = 2 √ making a − 1/a = 1 and a + 1/a = 5.30 shows the magnitude and phase function and Fig. then the delay time for low frequencies can be shown to be 2.. and so the transfer function is |Hc3 (jω)| = with = 1/2. (8. 8. Since θ(f) is the negative of the angle of the denominator polynomial.5 sec. delay.5 sec. Then the roots in the left half plane are at √ π π 1 5 sin + j cos 2 6 2 6 √ 1 15 = − +j 4 4 π 1 1 s1 = − sin = − 2 2 √ 2 1 15 s2 = s ∗ = − − j 0 4 4 s0 = − Then the denominator of the transfer function is √ √ 1 1 15 15 1 )(s + + j ) (s + )(s + − j 2 4 4 4 4 1 s 5s 1 = (s + )(s2 + + 1) = s 3 + s2 + + 2 2 4 2 Then H(s) has to be H(s) = 2 2s3 + 2s 2 + 5s + 1 4 where the denominator polynomial had to be multiplied by two to make the constant term unity. Fig. 2 Solution: Since C 3(ω) = 4ω3 − 3ω.5 Choose = 1/2 and determine the magnitude phase and step response of a third order Chebyshev low-pass filter. then ½ θ(f ) = − tan −1 2ω + tan−1 ω 2(1 − ω2 ) ¾ If θ(f ) is differentiated with respect to ω and ω set equal to zero. 165 .Example 8.

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Let fH be the positive solution of f − f02 = fc f and fL the positive solution f − f02 = −fc f a) Find fH and fL in terms of f0 and fc b) Show that fH − fL = fc 167 . but nothing else can be altered at will {b is not equal to 0 of course } and xm (t) is 1KHz sinusoid.4 The concerned regulatory agency has decreed that the maximum frequency deviation for F M stations will be 75KHz. and the behaviour of the new circuit at the frequency f given by f − f02 ± fc f must be the same as the behaviour of the old at fc = ±f. then what range will β have? b) What bandwidth will this require? c) If an AM − SSB with vestigial carrier were used instead of F M for the same signals as in (a) what bandwidth will be required? What is the ratio of the F M bandwidth to the AM − SSB bandwidth? 8.1 only the lower sideband is retained when x(t) = cos 220πt + cos 440πt is multiplied by cos(2πx106 t). then multiply by cos{106 − 110)t} to detect the AM − SC wave.3 The signal Xm (t) in Fig.5 For the low-pass to bandpass conversion let us suppose that the lowpass filter has a cutoff frequency at fc. 8. Multiply this signal by cos(2πx106t) to produce AM − SC . 8.1 Let x(t) = cos 220πt+ cos 440πt. a) If the maximum and minimum modulation frequencies it is desired to transmit are 15K Hz and 20Hz. 8. What will the output be now when signal is multiplied by cos{2π(106 − 110)t} and the frequencies above audio range are filtered out.73x107 t + bxm (t)} (a) What is the smallest bandwidth the phase modulated signal can have assuming that b. (b) If Xm (t) = sin377t and b = 1000. Since the alteration of the circuit is equivalent to replacing f by f − f02 /f the behaviour of the new circuit at f0 will be same as that of the old at f = 0. 8.32 is phase modulated and the output is x(t) = A cos{6.2 Suppose that in Prob.PROBLEMS 8. what is the output signal? This will indicate what happens when music is received by an AM − SC receiver with drifting oscillator. what is the maximum instantaneous frequency deviation? What is the carrier frequency? What is the bandwidth occupied by the signal? 8. Thus only these frequencies whose magnitude are less than 1M Hz are retained. Assuming that all but the audio frequencies are filtered out in the last step.

8. 8.9 but this time R = 2 in Fig.9 Show that the circuit of Fig. 8. This has a nominal impedence level of one ohm and a cutoff frequency of 1/(2π)Hz a) Design a high pass constant k filter with impedence level 5K and cutoff frequency of 30Hz b) Design a bandpass filter with bandwidth 10KHz. Show in particular that with this resistance level the filter is a third order Butterworth.34 is terminated in one Ohm and driven by a source with a one Ohm internal impedence.35 is a second order Butterworth filter. Does it have overshoot? = 3/4.33. 8. 8.10 Show that the circuit appearing in Fig.34 and show that this is now a third order Chebyshev filter with = 1/2 8. as shown in Fig.2 c) Show that fH − fL = f0 8. then find the transfer function of the filter.34 with R = 1.7 If the constant k filter of Fig.6 The constant k low-pass filter is shown in Fig. 8. 8. Does it have overshoot? 8. Find and Plot carefully its response.36 is Chebyshev second order filter with Find its step response. 168 . 8. centre frequency 80KHz and 10K impedence level.8 Repeat Problem 8.

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analysis of seismic signals.δT (t) (9. with a sampling interval T seconds (Fig. Eq. the output of the sampler can be considered to be a train of impulses with their height proportional to x(t) at the sampling instant (Fig.2) The value of x(t) is needed only at t = nT and furthermore for a physical system x(t) = 0 for t < 0. Taking the Laplace transform of x∗ (t) directly from Eq. and the associated theory of discrete time systems can often be employed in a number of science and technology disciplines. it is seen that the switch output is a train of finite width pulses.1) where δ(t) is the unit impulse function occurring at t = 0.CHAPTER .IX Z-TRANSFORM 9.2) The ideal sampling function δT (t) represents a train of unit impulses.1 INTRODUCTION Digital signal processing has become an established method of dealing with electrical waveforms. T . 9. The signals in sampled data system may be of the form of a periodic or an aperiodic pulse train with no information transmitted between two consecutive pulses. Typical applications of this technique are analysis of biomedical signals. τ seconds. vibration analysis. and δ(t− nT ) is a delayed impulse function occurring at t = nT . However.3) X ∗ (s) = L[x∗ (t)] = ∞ X n=0 x(nT )L [δ(t − nT )] (9.1). This train of pulses may be natural or man made through some sampling process.3) x∗ (t) = n=0 Thus we see that x∗ (t) is a weighted sum of shifted unit impulses.5) 171 . picture processing. Referring to Fig. 9. is negligible compared with the interval between successive samples. x(t). if the pulse width. Therefore x∗ (t) = x(t). to be sampled by a switch closing periodically for a short time. therefore ∞ X x(nT )δ(t − nT ) (9. and is defined as δT (t) = ∞ X n=−∞ δ(t − nT ) (9.4) becomes X ∗ (s) = Σ∞ x(nT )e −nT s n=0 (9. (9. 9. τ . A simple but adequate model of the sampling process is one which considers a continuous input signal.1.4) Since the Laplace transform of the unit impulse δ(t − nT ) is e−nTs . (9. speech analysis and sampled data control systems.

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In contrast if ωs/2 is not greater than the highest frequency component in the continuous signal (Fig. then the original signal can theoretically be recovered from the spectra of x ∗ (t) (Fig.2 that for the impulse modulator.4) that as a result of impulse sampling the frequency spectrum of x(t) is repeated infinitum at intervals of jw. 9. then the folding of frequency response function occurs. It has been established that the sampled data signal has infinite number of complementary frequency spectra. 9. x∗ (t) = δT (t)x(t). Let us now consider the frequency spectra of X ∗ (t). which means that there must be an infinite number of associated pole zero patterns in its s-plane representation. and consequently the original signal cannot be reclaimed from the sampled data signal. 9. so that any point sx in the s-plane transforms to a corresponding point zx in the z-plane as shown in Fig. (9. that is ∞ X C n ejnωst δT (t) = n=−∞ 1 T δT (t)e−jnωst dt T 0 and ωs is the sampling frequency equal to 2π/T rad/sec.3b). Consequently. then Z T δt(t)e− jnωst dt = 1 Cn = 0 1 and therefore Cn = T . and it has properties that enable linear difference equations to be solved using straight forward algebraic manipulations.3c). fortunately.3.7) Thus we see from Eq. we obtain ∞ 1 X X ∗ (s) = L[x∗ (t)] = X(s − j nωs) T n= −∞ therefore X ∗ (jω) = ∞ 1 X X[j(ω − nωs)] T n=− ∞ We have seen in Fig. 9. therefore ∞ 1 X x∗ (t) = x(t)e jnω st (9.3a).We can also expand Eq. which may be avoided by increasing the sampling frequency. Referring to Fig.1) as Fourier series. Suppose that we let z = eS T = e(σ +jω)T . if ωs/2 is greater than the highest frequency component of x(t) (Fig. hence where Z δT (t) = Taking Laplace transform and using the associated shifting theorem. 9.2 THE Z-TRANSFORM The z-transform is simply a rule that converts a sequence of numbers into a function of the complex variable z. Since the area of an impulse is unity. (9. 9. the analysis of any sampled data signal or system is extremely difficult when working in the s-plane. However. 9. it is possible to use Z-transfrom instead. which gives a good mathematical description.4. then |z| = e and 6 z = ωT . σT 173 . The errors caused by the folding of the frequency spectra are generally referred to as aliasing errors.6) T n=−∞ ∞ 1 X jnωst e T n= −∞ (9.

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also has a z-transform for the sampled function.8) In general. Let us consider Eq. When is negative |z| < 1 and when is positive |z| > 1. 9.1. |z| > e −aT (9. Hence a strip ωs wide in the left hand half of the s-plane transforms to the area inside the unit circle in the z-plane (Fig. which possesses Laplace transform.5) X ∗ (s) = ∞ X n=0 x(nT )e− nTs Since z = esT . find X (z) for sampling period T . Solution: From Eq.5) X ∗ (s) = = substituting z = esT .11) (9.12) .5). the above equation can be written as X(z) = ∞ X n=0 x(nT )z −n (9. Example 9.1 Let x(t) = e−at . |e −(s + a)T | < 1 1 1 − e−aTz −1 . jω 0 ωs 8 ω s 4 3ω s ω8 s ω 2 5ω s 8 3ω s 4 7ω s 8 s ωs = 2π/T z = 16 ωT 16 0 16 45o 16 90o 16 135o 16 180o 16 225o 16 270o 16 315o 16 360o The most important effect of z-transformation is that since the poles and zeros of x∗ (t) are spaced at intervals of ωs = 2π/T rad/sec in the jω direction.Referring to Table 1. (9. ∞ X n=0 x(nT ) = e−an T e−ante−nT s 1 . (9. it is seen that the imaginary axis in the s-plane transform to the P P circumference of the unit circle in z-plane.1 σ = 0.10) 1 − e− (s + a) T [X ∗ (s)] z = esT = X (z) = z z − e−a T 175 (9. all sets of poles and zeros in the s-plane transform to a single set poles and zeros in the z-plane. any continuous function. Table 9.9) (9.

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|z| > 1 j2 z − e jωT z − e−jωT ∙ ¸ jωT − jωT z e −e = j2 z 2 − (e jωT + e− jωT )z + 1 X (z) = z sin ωT z 2 − 2z cos ωT + 1 (9.19) .5) and Eq.18) Find the corresponding z-transform.16) therefore X(z) = ¸ ∙ 1 1 z . (9. ∞ X ∙ ejnωT − ejnωT ¸ Z −n j2 n=0 # "∞ ∞ X X jnωT −n −jnωT −n = (1/2j) (e )z − (e )z n=0 n= 0 X(z) = (9. (9. + .14) now similarly ∞ X (ejnωT )z − n = n=0 ∞ X (e−jnωT )z − n = n=0 z . for |z| > 1 z − e−jωT (9.2 Suppose that input signal of a digital filter is x(t) = sin ωt what is the z-transform of x∗ (t)? Solution: x∗ (nT ) = sin ωnT = (e jnωT − e− jnωT )/j 2. Solution: X(s) = Using partial fraction method. b− a ∙ + a a− b s+ b s ¸ 1 1 1 + − X(s) = a−b s+a s+b X(s) = 177 1 (s + a)(s + b) (9.15) (9. therefore from Eq.Example 9.13) (9. f or |z| > 1 z − ejωT z . 1 1 1 1 .8).17) Example 9.3 Suppose that the transfer function of a system is X(s) = 1 (s + a)(s + b) (9.

2 Table of Z .20) Table 9.Now from Table 9.2 X(z) = ¸ ∙ 1 z 1 − + a−b z − e− aT z − e−bT −bT −aT − e )/a − b z(e = (z − e− at(z − e−bT ) (9.Transforms Laplace Transform 1 1 s 1 1 − e−T s 1 s2 1 s3 1 sn+1 1 s+ a 1 (s + a) 2 a s(s + a) ω s2 + ω2 ω (s + a) 2 + ω2 s s2 + ω2 s+a (s + a) 2 + ω2 Time Function unit impulse δ(t) unit stepu(t) δT (t) = t t2 2 tn n! e− at te−at 1 − e−aT sin ωt e− aT sin ωt cos ωt e− at cos ωt P∞ n=0 δ(t − nT ) Z-Transform 1 z z−1 z z−1 Tz (z − 1)2 T 2 z(z + 1) 2(z − 1)3 z (−n)n ∂ n lims→0 ( ) n! ∂a n z − e−aT z z − e −aT T ze −aT (z − e −aT )2 (1 − e− aT )z (z − 1)(z − e−aT ) z sin ωT z 2 − 2z cos ωT + 1 ze−aT sin ωT −2 aT − 2zω − aT cos ωT + 1 ze z(z − cos ωT ) z 2 − 2z cos ωt + 1 z 2 − ze−aT cos ωT z 2 − 2ze− aT cos ωt + e−2at 178 .

Power Series Expansion Expanding X(z) into a power series in z − 1 by long division.22) From the z-transform table (Table 9. in the region of convergence of X(z). 2. (9. The z-transform is manipulated into partial fraction expression and the z-transform table is used to find the corresponding time function. The value of x(t) at the sampling instant t = nT can be obtained by the following formula: x(nT ) = 1 2πj I X (z)z n−1 dz (9.24) hence x(nT )δ(t − nT ) (9. The time function x(t) may be obtained from X (z) by the inversion integral. . + (1 − e− naT )z− n + ..9. This can be accomplished by one of the following methods: 1. it is often desirable to obtain the time domain response from the z-transform.e. . X(z) = (1 − e− aT )z −1 + (1 − e− 2aT )z −2 + (1 − e− 3aT )z −3 + .3 THE INVERSE Z-TRANSFORM Just as in the Laplace transform method. Partial Fraction Expansion Method Equation (9. . the corresponding time function at the sampling instant is x(nT ) = 1 − e−anT x∗ (t) = = ∞ X n=0 ∞ (9.23) (1 − e −aT )z (z − 1)(z − e− at ) (9.25) X (1 − e−anT )δ(t − nT ) n=0 2. The coefficient of z −n corresponds to the value of time function x(t) at the nth sampling instant.22) may be written as X(z) = z z − z − 1 z − e −aT (9. It may be emphasized that only the value of x(t) at the sampling instants can be obtained from X(z). since X(z) does not contain any information on x(t) between sampling instants.4 Given the z-transform X(z) = find the inverse z-transform x∗ (t). and c is of such a value that all the poles of X(z) are enclosed by the circle. The z-transform signal X(z) is expanded into power series in powers of z −1 . 3.21) Γ where Γ is a circle of radius z = ecT centered at the origin in the z-plane. 1. .2). Example 9. i.26) 179 .

+ (1 − e− ant )δ(t − nT ) + . .Correspondingly X ∗ (t) = 0xδ(t) + (1 − e−aT )δ(t − T ) + (1 − e− 2aT )δ(t − 2T ) + . Real Inversion Integral Method From Eq. 2.30) . .28) 1. .29) The region of convergence is at least the intersection of regions of convergence of z[f1 ] and z[f2 ]. Shifting Theorem (Real Translation) If Z[f] = F (z) then where n is an integer Z[f(t ± nT )] = z ±n [F (z)] 180 (9. Thus Z is a linear operator on the space of all z-transformable functions f (nT ) for n = 0. (9. . 2.TRANSFORMS. Linearity of the z-Transform For all constants C1 and C 2 . . the following property holds: ∞ X n= 0 Z(C1 f1 + C2 f2 ) = [C 1 f1 (nT ) + C2 f2 (nT )] z −n f1 (nT )z −n + C 2 ∞ X n=0 = C1 = C1 Z(f1 ) + C2 Z(f2 ) ∞ X n=0 f2(nT )z − n (9.27) n=0 3. (9. . .21) we have x(nT ) = at poles of X (z) 1 2πj I X(z)z n− 1 dz = Γ X Residue of X(z) z n−1 ¯ ¯ −at n¯ (1 − e −at)z n ¯ ¯ + (1 − e )z ¯ ¯ z − e− at ¯ z=1 z−1 z= e−at −anT =1−e 9. 1. ∞ X = (1 − e− anT )δ(t − nT ) (9.4 SOME IMPORTANT THEOREMS OF Z.

then F (z) = Z[t] = From Theorem 3 Tz (z − 1)2 ¤ £ Z e±aT f (t) = F (ze±aT ) (9.37) 181 .32) Proof: By definition (9. we can easily obtain the z-transform of the forward difference as well as the backward differences.Proof: By definition z[f(t ± nT )] = = ∞ X X k= 0 n=0 ∞ f (kT ± nT )z −k f (kT ± nT )z −(k± n) . Following a similar procedure.5 Apply the complex translation theorem to find the z-transform of te Solution: If we let f (t) = t. Eq. ∞ ¤ X £ e(nT )z − n = F (z 1 ) Z e±aT = 1 n=0 (9.33) If we let z1 = ze ±aT . (9.36) Z [te−at ] = F (ze−at) = T (z −at) (ze−aT − 12 ) T ze −aT = (z − e −aT )2 (9.z ±n ∞ X k=0 = z ±n f(kT ± nT )z − (k±n) (9. 3.35) − at (9.34) Example 9. Complex Translation £ ¤ ¤ £ Z e ±aT f(t) = [F (s ± a)] = F ze ±aT ∞ £ ¤ X Z e± aT f(t) = f (nT )e ±anT z −n n=0 (9.31) = z ±n F (z) This Theorem is very useful in the solution of difference equations.32) becomes hence.

45) .208 (9.0416z + 0. " ∞ X k=0 ∞ X k=0 F1 (z)F 2 (z) = Z Proof: By definition f1 (kT )f2(n − k)T # (9.792z z−1 ) z (z 2 − 0.39) Example 9.416 + 0.6 Given F (z) = 0.0416z + 0.208) 2 (9. then t→∞ lim f ∗ (t) = lim(1 − z − 1 )F (z) z →1 (9. 6.792z 2 (z − 1)(z 2 − 0. Initial Value Theorem If the function f(t) has the z-transform F (z). the final value of f ∗ (t) is unity. and limit of F (z) exists. and (1 − z −1 )F (z) has no poles on or outside the unit circle centered at the origin in the z-plane.208 (9. the initial value of f (t) is zero.43) F1 (z)F2 (z) = But we know that f1 (kT )z −k F2 (z) (9.44) z− k F2 (z) = Z [f2 (t − kT )] 182 (9.42) If f1 (t) and f2 (t) have the z-transform F1 (z) and F2 (z) then. Real Convolution Theorem 0. then lim f ∗ (t) = lim F (z) t→0 z→∞ (9.208) 0.41) Therefore.38) 5. Final Value Theorem If the function f(t) has the z-transform F (z). Final value of F (z): From Theorem 5 lim f ∗ (t) = lim(1 − z − 1 )F (z) t→∞ z →1 = lim( z →1 Therefore.792z = lim 2 =1 z →1 z − 0.4. Initial value of F (z): From theorem 4 lim f ∗ (t) = lim F (z) t→0 z →∞ = lim z →∞ =0 ∗ 0.792z 2 (z − 1)(z − 0.40) determine the initial and final value F (z).416z + 0.

52) 183 .50) Differentiation with respect to second independent variable Z[ ∂ ∂ f(t. Complex Differentiation (Multiplication by t) If F (z) is the z-transform of f . Second Independent variable limit value Z[lim f(t. a)] = lim F (z. then Z[tf ] = −T z Proof: By definition Z[tf] = ∞ X (nT )f (nT )z −n n=0 ∞ X n=0 d F (z) dz (9.49) The term in the bracket is a derivative with respect to z Z[tf] = − T z = − Tz = − Tz 8. a)] = F (z. a) ∂a ∂a (9.51) 9. a) a→a 0 a→a 0 (9.Hence F1 (z)F2 (z) = = = ∞ X X n=0 k=0 ∞ f1 (kT )Z[f2 (t − kT )] f1 (kT ) ∞ X n=0 (9.47) f1 (kT )f2 ((n − k)T ) z −n 7.48) = − Tz f (nT )(−nz −n−1 ) (9.46) k=0 "∞ ∞ X X k=0 f2 [(n − k)T ] z − n # (9. d X f (nT )z −n dz n=0 ∞ ∞ X n=0 f (nT ) d −n z dz d F (z) dz (9.

x(n − 1)T . x(0) (9. the output c(t) and c∗ (t) of the system is obtained by means of the principle of superposition.6a is given as C (s) (9. the output of the switch S2 to a unit-impulse input is ∞ X c∗ (t) = g ∗ (t) = c(nT )δ(t − nT ) (9. ∞ X G∗ (s) = g(nT )e−nT s n=0 (9.6b at t = 0. x(n − 1)T .59) x(nT )g(0)e 184 . Once the weighing sequence of a network G is defined. Integration with respect to second independent variable Z if the integral is finite.10. 9. P Assume that S1 is an ideal sampler so that x∗ (t) = n x(nt)δ(t − nt) ∙Z a a0 ¸ Z f(t. a)da = a F (z. 9. . . . x∗ (t). . 9. x(0)... 9. + x[(n −1)T ]g(T ) + x(nT )g(0) (9. a)da a0 (9. At the time t = nT . Suppose that an arbitrary function x(t) is applied to the system of Fig.7.58) Multiplying both sides of the last equation by e −nT s and taking the summation for n = 0 to n = ∞. . . Fig. The signals x(t). . x(n − 2)T. . . the sampled input to G is the sequence x(nT ).53) If a fictitious sampler S2 with the same sampling period T as that of S1 is placed at the output. we have ∞ ∞ ∞ X X X c(nT )e− nT s = x(0)g(nT )e −nT s + x(T )g(n − 1)T e− nT s n=0 n=0 n=0 +. c∗ (t) are illustrated in Fig. 9.54) G(s) = X(s) For a system with sampled-data.56) which is the pulse transfer function of system G. .57) or c(nT ) = x(0)g(nT ) + x(T )g[(n − 1)T ] + x(2T )g[(n −2)T ]+ .55) n=0 where c(nT ) = g(nT ) is defined as the ”weighting sequence” of G. the output sample c(nT ) is the sum of the effects of all samples x(nT ).5 THE PULSE TRANSFER FUNCTION The transfer function of the open-loop system in Fig. c(t).6b illustrates a network G which is connected to a sampler S with sampling period T. that is X c(nT ) = effects of all samples x(nT ). + + ∞ X n=0 ∞ X n=0 x [(n − 1)T ] g(T )e−nTs + −nT s ∞ X n= 0 x(nT )g(0)e−nT s (9.

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Z-Transform of Cascaded Elements with Sampling Switches between them (9. C ∗ (s) = G∗ (s)G∗ (s)X ∗ (s) 2 1 The z-transform of the above equation is C(z) = G1 (z)G2 (z)X(z) 2.69) (9. Taking the z-transform of both sides of Eq.70) . .or ∞ X n= 0 £ ¤ c(nT )e−nT s = x(0) + x(T )e− Ts + x(2T )e− 2T s + . (9.65). . (9. (9.67) ∗ (9. (9.62) where G∗ (s) is defined as the pulsed transfer function of G and is given by Eq. 9.63) Fig. The z-transform relation between the output and the input signals is derived as follows.56). we have C(s) = G2 (s)G ∗ (s)X ∗ (s) 1 Taking the pulsed transform of the last equation.64) (9.66) Z-Transform of cascaded elements with No sampling switch between them Fig.6 Z-TRANSFORM OF SYSTEMS 1. 9. The output signal of G 1 is D(s) = G 1(s)X(s) and the system output is C (s) = G2 (s)D∗ (s) Taking the pulsed transform of Eq. ∞ X n=0 g(nT )e−nT s ∞ X n=0 (9.60) from which or simply ∞ X n=0 c(nT )e−nT s = ∞ X n=0 x(nT )e −nT s g(nT )e− nT s (9. The z-transform relation of output and input is derived as follows: The transform of the continuous output is C(s) = G1 (s)G 2 (s)X ∗ (s) 186 (9. The two elements are separated by a second sampling switch S which is synchronized to S1 .8 a illustrates a sampled data system with cascaded elements G1 and G2 .8 b illustrates a sampled data system with two cascaded elements with no sampler between them.64) yields D ∗ (s) = G∗ (s)X ∗ (s) and substituting D (s) in Eq.65) (9.68) (9.61) C ∗ (s) = X ∗ (s)G ∗ (s) (9. (9.62) yields C(z) = X (z)G(z) 9. we have.

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79) .71) is G1 G∗ (s) 6= G∗ (s)G∗ (s) 2 1 2 C(z) = G 1G 2 (z)X(z) (9.77) (9. For the system shown in Fig.The pulsed transform of the output is C ∗ (s) = G 1 G∗ (s)X ∗ (s) 2 where G1 G ∗ (s) = [G1 (s)G 2 (s)] = 2 In general. (9.75) C(z) = G 1G 2 (z)X(z) ¸ ∙ a X(z) = s(s + a) z(1 − e−aT ) z = × (z − 1)(z − e−aT ) z − 1 z 2 (1 − e−aT ) = (z − 1)2 (z − e−aT ) 3. and x(t) is a unit step function. General Closed Loop Systems (9. Solution: The output of the system in case ’a’ is C(z) = G1 (z)G 2 (z)X(z) z az z = × × z − 1 z − e− aT z −1 az 3 = 2 (z − e −aT ) (z − 1) The output in case 0 b0 is (9. if G1 (s) = 1/s.78) (9. Find C(z) in both the cases. 9.74) Example 9.72) The z-transform of the Eq.8 a and b.7 For the sampled data system in Fig. ∗ ∞ 1 X G1 (s + jnωs )G2 (s + jnωs ) T n=− ∞ (9.9 the output transform is C(s) = G(s)E ∗ (s) The Laplace Transform of continuous error function is E(s) = X(s) − C (s)H(s) or E(s) = X(s) − H(s)G(s)E∗ (s) 188 (9.73) (9.76) The transfer function of a closed loop sampled data system can also be obtained by the procedure in the last sections.71) (9. G(s) = a/(s + a). 9.

8 STABILITY ANALYSIS A sampled-data system is considered to be stable if the sampled output is bounded when bounded input is applied.84) G∗ (s) 1 + H G∗ (s) (9. otherwise the system response obtained by the z-transform method is unrealistic or even incorrect. The derivation of z-transform is based on the assumption that the sampled signal is approximated by a train of impulses whose areas are equal to the input time function of the sampler at the sampling instants. (9. This assumption is considered to be valid only if the sampling duration is small. 2. Therefore. we have E∗ (s) = X ∗ (s) − HG ∗ (s)E ∗ (s) from which E(s) = X ∗ (s) 1 + HG∗ (s) (9. However. (9. for any C (z). there may be hidden oscillations between sampling instants. it is necessary that the transfer function G(s) must have at least two more poles than zeros [or g(t) must not have a jump at t = 0]. However.Taking the pulsed transform of the last equation. the inverse transform c(nT ) describes c(t) only at the sampling instants t = nT . The closed loop transfer function of the sampled-data system in Fig.9 is given as C ∗ (s) G∗ (s) = X ∗ (s) 1 + H G∗ (s) (9. C(s) = The pulsed-transform of c∗ (t) is C ∗ (s) = G∗ (s)E∗ (s) = Hence the z-transform of c(t) is C(z) = G(z) X(z) 1 + HG(z) (9.83) G(s) X ∗ (s) 1 + HG∗ (s) (9.77). 1.81) The output transform C(s) is obtained by substituting E ∗ (s) from Eq. which may be studied by special methods. 9.82) 9.7 LIMITATIONS OF THE Z-TRANSFORM METHOD We have seen that z-transform is a convenient tool for the treatment of discrete systems. 9. compared to the significant time constant of the system.80) (9. 3. The z-transform C (z) specifies only the values of the time function c(t) at the sampling instants.85) where 1 + HG∗ (s) = 0 is the characteristic equation of the system. The stability of the sampled data system is entirely determined by the location of the roots of the characteristic 189 .81) into Eq. it has certain limitations and in certain cases care must be taken in its applications. In analysing sampled data by z-transform method.

91) 190 .368z + 0.7z − 0. The stability of the sampled data system concerns the determination of the location of the roots of the characteristic equation with respect to the unit circle in the z-plane. 6 1 r = −31. (9.368) (9. r = σr + jwr . i.13 equation is -11.3 r2 −27− 44. the z-transform of G(s) is G(z) = 22. Since the right half of the s-plane is mapped into the exterior of the unit circle in the z-plane. (9.74 3. Specifically. We will not discuss all the stability techniques in detail. The following example illustrates how the modified Routh test is performed for a sampled data feedback system.e.3r 2 − 11. we get 14.368 = 0 r−1 (9. the Routh test may be performed on the polynomial in the variable r.74r + 3.57 G(s) = 2 (9.88) The characteristic equation of the system may be written as z 3 + 5.86) in the last equation yields ∙ r+1 r−1 ¸3 + 5. as shown in Fig.86) Where r is a complex variable.8 Let the open loop transfer function of a unity feedback system with sampled error signal be of the form 22.368 = 0 Substitution of Eq. In terms of the z-transform. therefore. The Routh-Hurwitz Criterion Applied to Sampled Data System.57z(0. 1.1 2.94 ∙ r +1 r−1 ¸2 + 7.3 0 r 3.7 ∙ ¸ r+1 − 0. r= or z= z+1 z−1 r+1 r−1 (9.13 0 (9. the characteristic equation of the system is written as 1 + HG(z) = 0.89) Simplifying Eq.90) (9. 9.13 = 0 The Routh tabulation of the last 14. A convenient method is to use bilinear transformation.94z 2 + 7. the stability requirement states that all the roots of the characteristic equation must lie inside the unit circle.equation. none of the roots of the characteristic equation must be found in the right-half of the s-plane.27 r3 2. Example 9. This transformation maps the interior of the unit circle in the z-plane into the left half of the r-plane.10. since such a root will yield exponentially growing time functions.264) (z − 1)2 (z − 0. but outline briefly only two methods namely Routh Hurwitz Criterion and Root Locus Method.27r3 + 2.90).87) s (s + 1) Solution: If the sampling period is 1 sec.

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which corresponds to two roots outside the unit circle in the z-plane.11a. z = 0.632kz (z − 1)(z − 0.The root loci must start at the poles (k = 0) and end at the zeros (k = ∞) of G(z).92) where H G(z) is a rational function in z. G(z) becomes G(z) = 0. If the sampling period is changed to T = 5 sec. Since the characteristic equation of a simple sampled data system may be represented by the form 1 + HG(Z) = 0 (9. in the construction of the root loci discussed in Chapter 5 is still valid.94) which has poles at z = 1 . 9. 192 . 9. and that in investigating the stability of sampled data system from the root locus plot. Example 9. If the sampling period T is 1 sec.33.93) Draw the root loci of the system for T = 1 sec and T = 5 sec. (9. The significant difference between the present case and the continuous case is that the root loci in Eq. The pole-zero configuration of G(z) is shown in Fig. and shows that the system is unstable. the unit circle rather than the imaginary axis in the z-plane should be observed.0067) (9. The Root Locus Technique The root locus technique used for analysis and design of continuous data system can also easily be adapted to the study of sampled data systems. G(z) becomes G(z) = 0. the root locus method may be applied directly to the last equation without modification. The complete root loci for T = 1 sec intersects with the unit circle occurs at z = −1 and the corresponding value of k at that point is 4.368 and a zero at the origin.368) (9. the openloop transfer function of the system is given as G(z) = kz(1 − e−T ) (z − 1)(z − e− T ) (9.Since there are two changes of sign in the first column of tabulation.95) The root loci for T = 5 sec are constructed in Fig.11b. the characteristic equation has two roots in the right half of the r-plane.33 for T = 1 sec.933kz (z − 1)(z − 0. Solution: The characteristic equation of the system is 1 + G(z) = 0.02 as compared to the marginal k of 4.92) are constructed in the z-plane. It is clear that. The following example shows that construction of root loci for a sampled data system.9 Consider a unity feedback control system with sampled error signal. whose root loci are to be determined when k is varied from 0 to ∞ . The marginal value of k for T = 5 sec is found to be 2. 2.

5z (z − 1)(z − 0.2. 0.5 A digital filter has a pulse transfer function G(z) = Determine: z2 − 0. (c) power series expansion. (b) whether or not the filter is stable.05z − 0.4 Obtain the inverse z-transform of G(z) = 0. (c) a general expression for the filter’s impulse response.9. (b) the partial fraction expansion.5) z(z 2 + 2z + 1) (z − z + 1)(z 2 + z + 1) 2 9.08.PROBLEMS 9.4} (d) (1/4) 4 for n > 0 for n > 0 9. of the filter’s poles and zeros.05 z 2 + 0. −1. (a) x(t) = te −αt (b) x(t) = e−at sin ωt (a=constant) 9. Determine the sampler output x∗ (t) and evaluate the pulsed transform X ∗ (s) by the Laplace Transform method. 9. 8. 193 . (a) the real inversion formula. (d) the filter’s linear difference equation.2 Derive the z-transform for the following functions (a) (b) 1 s3 (s + 2) 1 s(s + s)2 (c) {2.5.3 Evaluate the inverse z-transform of G(z) = by the following methods.2 (a) the location in the z-plane. (e) initial and final values of the output of the filter for a unit step input. −0.1z − 0.1 The following signals are sampled by an ideal sampler with sampling period T .

s(s2 + 100) 9. Determine the marginal value of k for stability in each case.8 Obtain the initial and final value of the following functions.9.7 The sampled data system shown below has a transfer function G(s) = K s(1 + 0. Determine the stability of these systems. 9. 194 .5z −1 ) 100 T = 0.2s) Sketch the root locus diagram for the system for T = 1sec and 5 sec.6 The characteristic equation of certain sampled data systems are as follows. (a) G(z) = (b) G(z) = 2 (1 − z − 1 )(1 − 0. (a) z 3 + 5z 2 + 3z + z 2 = 0 (b) 3z 5 + 4z 4 + z 3 + 2z 2 + 5z + 1 = 0 (c) z 3 − 1.9 For the open-loop sampled data system given below G(s) = Use the z-transform method to evaluate the output response.5z 2 − 2z + 3 = 0 9.2z −1 ) 1 (1 − z − 1 )(1 − 0. x(t) = unit step.1 sec.

we shall apply z-transform method (generating function method) to the solution of certain type of linear difference equations.1).1.1) 2Ri0 − Ri1 = V i1 = 2i0 − V R Substituting the value of i 1 in Eq.1) −I(z) + 3zI(z) − 3zi 0 − z 2 I(z) + z 2 i 0 + zi1 = 0 or or (1 − 3z + z 2 )I(z) = z(zi0 − 3i0 + i1 ) I(z) = From 0 00 Loop or z(zi 0 − 3i0 + i 1 ) z2 − 3z + 1 (10.1 Z-TRANSFORM METHOD FOR SOLUTION OF LINEAR DIFFERENCE EQUATIONS Several methods. 10. we formulate the equation of the current in any loop of a ladder network and find its solution by z-transform method.3) (10. These equations are usually encountered in physical. we get 195 . In the classical approach. It may be mentioned that the application field of z-transform is not limited to the above areas and with the introduction of digital computers in control and instrumentation. the matrix method. such as the classical method. (10. 10. 1. Eq. we could set up the (k + 1) loop equations and solve for in which would be a cumbersome process. together with end condition is sufficient to describe the network. since the network is the repetitive structure and all loops except two ends are alike. the recurrence method and the transform method exist for the solution of difference equations. (10. Assume that resistances except RL are of the same value R. we will demonstrate its usefulness in the analysis and design of networks. In this section. (10. However. In the following. by z-transform method. Applying z-transform to Eq.1) is true for any n except 000 and 0 k 0 .2) (10. economic and physiological systems. its scope has become almost unlimited. (10. 2.2). we could formulate one loop equation and two terminal equations. The formulation of the difference equation can be expressed in several forms such as backward or forward method or the translational form. Suppose that it is required to find the current in the nth loop. we make the following observations. The equation for the (n + 1)th loop is −Ri n + 3Rin+1 − Ri n+2 = 0 Instead of writing down the other K equations. In the following sections. Eq.4) (10.CHAPTER X APPLICATIONS OF Z-TRANSFORM The method of z-transform is an efficient tool for dealing with the linear difference equations. Consider the ladder network in Fig. sampled data control systems and digital filters.

As the detailed discussion of control theory is beyond the scope of this book and the fact that we intend to demonstrate the application of z-transform in this field.5) From the tables of inverse z-transform. no simple rules are available for the determination of appropriate compensation networks from the root locus diagram. The root locus plots which are plotted in the z-plane have quite similar properties to those of the root locus for continuous systems in s-plane.2. However. Nyquist plots. (10. Therefore.2 SAMPLED DATA CONTROL SYSTEM DESIGN IN THE Z-PLANE Design and synthesis of sampled data control systems is a subject of control theory. The most elementary problem in the root locus design is the determination of the loop gain to yield suitable relative stability. Magnitude and Phase plots either by direct applications or after bilinear transformations. much knowledge concerning the transient response of the system can be obtained by observing the location of the roots for the particular loop gain k.6) 3 2 √ 5 sin hω0 = 2 t = nT = n for T = 1sec.1 Design in the z-Plane by using the Root Locus Method. 196 . The loop gain of the system can be adjusted to give appropriate dynamic performance as measured by the position of the complex poles with respect to the constant damping ratio curves inside the unit circle.I (z) = = z[zi 0 − 3i0 + 2i0 − V ] R z 2 − 3z + 1 h i z z − (1 + RV i 0) i0 z 2 − 3z + 1 (10. Once the root loci of the characteristic equation are plotted in the z-plane. This method is chosen because of the fact that it was thoroughly discussed in Chapter 5 for continuous systems and the reader will not find any difficulty in following its application for sampled data systems. and methods such as Bode plots. we readily obtain in as follows. Nyquist plots.8) The value of i 0 can be found by substitution Eq. 10. Magnitude and Phase plots and Root Locus plots are generally employed and synthesis of control systems is carried out in the z-plane. cos hω0 = (10. However.6) into the equation of the end loop and solving for i0 . 10. it may be pointed out that z-transform method is equally useful and applicable in the design and synthesis by Bode Plots. " 1 2 i n = Z [I (z)] = i0 cos hω0 n + Where and −1 − √ V Ri 0 5 2 sin hω0 n # (10. the desirable characteristic equation roots are found by reshaping the root loci of the original system through adjustment of the loop gain and the use of the compensation networks. In terms of root locus design. we shall limit this section to the design and synthesis by root locus method only. the design in the z-plane with root locus usually involves a certain amount of trial and error.

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In the design of continuous data systems, usually, the design may fall into one of the following categories: (1) phase-lead compensation (2) phase-lag compensation. 10.2.2 Phase-Lead Compensation A simple phase-lead model on the ω-domain is described by the transfer function Dω = 1 + aτω (a > 1) 1 + τω ω= z −1 z +1 (10.9) (10.10)

where τ is a constant greater than or equal to zero. This transfer function produces positive phase shift that may be added to the system phase shift in the vicinity of the gain croos-over frequency (ωω ) to increase the phase margin. The pole-zero configuration of Eq. (10.9) is shown in Fig. 10.2.a. Note that the poles and zero of D(ω) always lie on the negative real axis in the ω-plane with the zero to the right of the pole. Substitution of ω = z − 1/z + 1 into Eq. (10.9) yields ∙ ¸ aτ + 1 z + 1−aτ 1+aτ D(z) = (10.11) τ + 1 z + 1 −τ 1 +τ Since τ and a are both positive numbers and since a > 1, the poles and zero of D(z) always lie on the real axis on or inside the unit circle in the z-plane; the zero is always to the right of the pole. A typical set of pole zero configuration of D(z) is shown in Fig. 10.2b. Illustrative example is represented in the following. Example 10.1 A sampled data feedback control system with digital compensation is shown in Fig. 10.3. The controlled process of the system is described by the transfer function G1 (s) = k s(s + 1) (10.12)

The root locus diagram of the uncompensated system is plotted in Fig. 10.4.

The sampling period is one sec. The open-loop transfer function of the system without compensation is 0.386k(z + 0.717) G h0 G1 (z) = (10.13) (z − 1)(z − 0.386)

Note that the complex conjugate part of the root loci is a circle with centre at z = 0.717 and a radius of 1.37. The closed-loop system becomes unstable for all values of K greater than 2.43. Let us assume that k is set at this marginal value so that the two characteristic equation roots are on the unit circle as shown in Fig. 10.4. Suppose that the transfer function D(z) of the digital controller is of the form ∙ ¸ 1−aτ aτ + 1 z + 1+aτ D(z) = τ + 1 z + 1 −τ 1 +τ where for phase-lead compensation, a > 1 and ∞ > τ > 0.
198

(10.14)

The constant factor (aτ + 1)/(τ + 1) in D(z) is necessary, since the insertion of the digital controller should not effect the velocity error constant kv while improving the stability of the system. In other words, D(z) must satisfy the condition lim D(z) = 1
z→1

(10.15)

The design problem now essentially involves the determination of appropriate value of a and τ , so that the system is stabilized. However, at this point, how the values of a and τ should be chosen is not clear. Although, we know from the properties of the root loci that an added open loop zero has the effect of pulling root loci toward it, whereas an additional open loop pole has the tendency to push the loci away. However, no simple ways exist for telling which combinations are the most effective for stabilizing the system because of the unlimited number of possible combinations of pole and zero of D(z). Several sets of values of a and τ are used in the following to illustrate the effects of phase-lead compensation. As a first trial, let a = 6.06 and τ = 0.165. The transfer function of the digital controller reads. D(z) = 1.72 z z + 0.717 (10.16)

and the open loop transfer function of the compensated system is D(z)Gh0 G 1 (z) = 0.64kz (z − 1)(z − 0.368) (10.17)

The root loci of the compensated system are shown in Fig. 10.5 as loci (2). It may be seen that for k = 2.43, one of the roots of the characteristic equation is on the negative real axis outside the unit circle, and the system is unstable. This shows that for the values chosen for a and τ , the compensated system is worse than the original system. From other sets of values for a and τ are tried, and the corresponding root loci of the compensated systems are plotted in Fig. 10.5 (only the positive complex conjugate parts of the root loci are shown). The characteristic equation roots of the compensated system when k = 2.43 are indicated on the loci. The pole and zero locations of D(z) of the various compensations are tabulated in Table 10.1. From the root locus diagram, we see that among the five compensation only when a = 2, τ = 0.4 and a = 3, τ = 0.1 result in stable systems. But then the damping ratios are less than 10 percent, which means that overshoots would exceed 70 percent, which is not acceptable. The general ineffectiveness of the phase-lead compensation is anticipated in this problem, since the original system is on the verge of instability and the situation is one for which phaselead compensation is not recommended. In the next section, we shall see that a phase-lag compensation is more satisfactory for improving the stability of this system. Table 10.1 Pole and Zero of D(z) = τ 0.1 0.165 0.4 1.0 3.0 a 3 6.06 2 2 3 £ aτ +1 ¤ z+ 1−aτ 1+aτ
τ +1 z + 1−τ 1+τ

zero of D(z) 0.538 0 -0.111 0.333 0.80

Pole of D(z) -0.818 -0.717 -0.375 0 0.50

200

5.Lag Compensation A simple phase-lag model is given by the transfer function D(ω) = where 0 < a < 1 and 0 < τ < ∞ The Pole-Zero configuration of the phase-lag D(ω) and D(z) are depicted in Fig. preferably 1/5 to allow the dipole to contribute a slight phase lag near the new gain cross-over. the constant ratio a of the integrating dipole should be chosen to be at least 1/4. the phase-lag compensation can be regarded as a means of increasing the velocity error constant kv by a ratio of 4.19) 1 + aτω 1 +τω (10. the complex conjugate parts of the original root loci are not affected significantly by the addition of the integrating dipole. Note that since a is less than unity.6. the root loci of the phase-lag compensated system are shown as loci (2). we note that the complex closed loop poles have a damping ratio of 60 percent when the loop gain is equal to 0.1.43. Substituting these values of a and τ into Eq.5 on the uncompensated loci.5) while keeping the complex loop poles relatively small. First. (10. which is the root locus of the original system.8. Note that the complex roots for K = 2. Referring to locus (1) in Fig. as shown in Fig. Therefore.43.5 to k = 2.10.0. Since τ is to be very large.3 Phase .43 on the compensated loci lie very close to the roots for K = 0.905 0. 10. The root loci of the compensated system when a = 0.98) (10.1.86 (from k = 0.717) (z − 1)(z − 0.20) In Fig.191 D(z)Gn0 G1 (z) = 0. 202 . In order to increase the velocity error constant by a factor of 4.18) (10.43).11) yields the transfer function of the phase-lag controller z − 0. The system is now to be stabilized by means of a simple lag compensator of Eq. which indicates that the value of τ should be large. (only the positive complex conjugate loci parts are shown). 10.2 Consider the same system given in Example 10. In essence.86(2. 10. For small values of τ. the poles and zero of D(z) will appear as an integrating dipole near the point z = +1.2.4 and 1. we investigate the effects of the phase-lag compensation on the root loci when small values of τ are chosen.905)(z + 0. and since a is less than unity. let a = 0.368)(z − 0.2 and τ = 50 are also plotted in Fig. (10. Example 10.5. for K = 2. with k = 2. 0. With a = 0.43/0. the pole is always to the right of the zero. as long as it is close to the z = +1 point.7.8. 10. Thus we. This shows that precise location of the dipole is not critical.07k(z − 0.86.18). the complex characteristic equation roots lie very close to those loci (2). the phase-lag compensation has made the system unstable.980 The open-loop transfer function of the compensated system is D(z) = 0. since from points on these loci the dipole and the pole of G h0 G1 (z) at z = 1 appear as a sample pole.8 (loci 3). 10. Let us assume that the design specification requires the damping ratio of the complex closed loop poles to be approximately 60 percent.2 and τ is chosen to be 100. the root loci of the system with phase-lag compensation are plotted for τ = 0.

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respectively. Example 10. It may be mentioned that once a low pass continuous filter is designed. it can readily be converted into bandpass or highpass continuous filter as already discussed in Chapter 8. is designed and subsequently this is transformed via an appropriate s-plane to z-plane mapping to give a corresponding digital filter pulse transfer function. y(n − 2)T . Butterworth and Chebyshev were discussed in Chapter 8. recursive and non-recursive filters.21) and |G(jω)| 2 = (10. and the derivation of G(z) is achieved by working directly in the z-plane. The mapping used in this section will be the bilinear z-transform though other transformation methods are also available. .the a0 s and b0 s of G(z). (1) Digital cutoff frequency. which are more economical in execution time and storage requirements as compared to non-recursive filters. we will present the design of recursive filters. The second method is a direct approach which is concerned with the z-plane representation of the digital filter. .g. which requires that a suitable prototype continuous filter transfer function G(s). namely x(n − 1)T .10. .3. The design of the recursive filter centres around finding the filter coefficients . G(z). y(n − 1)T. |G(jω)|2 = 1 ω 1 + ( ω c )2n 1 = | jω (Butterworth) 1 + (−1)ns 2n s= ωc 1 (Chebychev) 1 + 2 [Cn (ω)]2 (10. x(n − 2)T. The general equations for the Butterworth and Chebyshev filters are given below.22) where |G(jω)|2 is the squared magnitude of the filter’s transfer function. fcd = 100 Hz 204 . thereby yeilding a pulse transfer function which is a rational function in z.3 Derive the digital equivalent of a Butterworth low pass filter by bilinear transformation for the following specifications. The direct approach is used in the design of frequency sampling filters and filters based on squared magnitude functions. There are two main methods for the design of digital filters. In this section. In the following. is computed using the present inputs x(T ). . ωc is the cutoff frequency n is the order of the filter ω is the frequency is a real number and << 1 Cn (ω) is Chebechev polynomial The design of digital filter is carried out by using bilinear transformation to the continuous filter obtained for given specification.3 Z-TRANSFORM METHOD FOR THE DESIGN OF DIGITAL FILTERS There are two types of digital filters namely. The first method is an indirect approach. Recursive digital filters are more commonly referred to as infinite impulse response filters. . The term recursive intrinsically means that the output of the digital filter. we present one example of a low pass filter design. . .1 Indirect Approach Using Prototype Continuous Filter The continuous filters e. y(n)T . 10. and previous inputs and outputs.

65 (10.29) ¸ ∙ ¸ (z − 1) (z − 1) = 2000 (z + 1) (z + 1) ∙ (z − 1)2 (z + 1)2 ¸ (10.25) The analog cutoff frequency is obtained by the following equation ∙ ¸ ωcd T 2 ωca = tan T 2 ¸ ∙ 2 1X 10− 3 = tan 200πx 1X10− 3 2 = 650 rad/sec (10.(2) Sampling period T = 1 ms (3) Amplitude attenuation of 20 dB at 400 Hz. Obtain the amplitude response of G(z).24s + 422500 (10.24) As n has to be an integer.28) 422500 s 2 + 919.9956 = 2n log1 0 4 n = 1.26) (10. ∙ ¸ ω 2n ) 20dB = 10 Log1 0 1 + ( ωc ∙ ¸ ω 2n 2 = Log10 1 + ( ) ωc 99 = (4)2n 1. therefore n = 2 shall satisfy the filter specifications.27) The transformation from normalised low pass to low pass filter is achieved by submitting s/ωca for s in G(s) as follows G(s)pωt = £ 2 s 650 (the pre-warped transformed transfer function) G(s)pωt = For the bilinear z-transform s= And it follows that 2 T ∙ ¤ 1 + √ 2s 650 +1 (10.31) 205 .23) (10. The prototype continuous second order filter is given by G(s) = s2 + 1 √ 2s + 1 (10.30) s2 = 4x106 (10. Solution: The order of the filter is determined by the ratio of cutoff frequency and 20dB attenuation frequency as follows.

37) therefore (10.31) in Eq.16 jωT in Eq. Consider the magnitude squared function defined as |G(ejωT )|2 = 1 1 + [Fn (ωT )]2 (10.935 sin ωt) (10. such that on substitution z = e jωT an MIP in the z-plane results.82 cos 2ωT + j14.36) We need suitable trigonometric functions for [Fn (ωT )]2 .395ejωT + 6.116 cos 2ωT + jsni2ωT + 2 cos ωT + j25inωT + 1 = 14.38) .9 10.z 2 + 1) 6260980z2 − 715500z + 2584020 z 2 + 2z + 1 (10. These functions are known as ”mirror image polynomials” (MIPs). (10.935 sin ωT + 6. the other half being outside.3.35) can be obtained for T = 1 msec.35) The amplitude vs. When designing digital filters using this direct approach.34) or G(ejωT ) = A A= (cos 2ωT + 2 cos ωT + 1) + j(sn2ωT + 2 sin ωT ) (14.82sn2ωT − 16.935 cos ωt + 6. that is 1 ωT = (1 + cos ωT ) 2 2∙ ¸ 1 1 = 1 + (ejωT + e−jωT ) 2 2 ¸ ∙ 1 1 1 + (z + z −1 ) = 2 2 cos2 (z + 1)2 ωT = 2 4z 206 cos 2 (10.116 G(e jωT ) = j2 ωT (10.2 Direct Approach Using Squared Magnitude Functions A direct approach to the design of digital filters is to derive G(z) working in the z-plane. and is shown in Fig. (10. (10. One such function is cos(ωT /2). 10.29).82 sin 2ωT − 16.935 cos ωT − j16.116) + j(14.32) 14.82 cos 2ωt − 16.Substituting Eqs.30) and (10.935z + 6.82e . (10.32) The frequency response of G(z) can be obtained by substituting z = e which results G(z) = ej2 ωT + ejωT + 1 14. frequency for Eq. 422500 G(z) = (z− 2 4x106 (z+ 1)2 + 1838480 (z −1 ) + 422500 1) ( z+1) = or 422500(z + 2.33) (10.2 − 16. we obtain the digital filter pulse transfer function as follows. we seek functions that produce half of the poles within the unit circle.82z 2 − 16.

45) (10.47)n 1 + 0. . at ω = 400x2π p q + pn 1 + (9.similarly sin now consider 2 (z − 1)2 ωT = 2 −4z 1+ h 1 sin2 ( ωT ) 2 sin 2( ωc T ) 2 (10. we can find the corresponding factors in the z-plane by solving p = (z − 1)2 / − 4z. 2 . ω = 400x2π = (1 − cos ωT ) = (1 − cos 2 2 2 1000 q = sin 2 ∙ |G(z)| 2 = qn 1 1 1 £ h in = ¤n = = . Substituting z = ejωT yields |G(z)|2 = qn q + pn n i2 (10. as given in Eq.9045.40) (10.0955 2 2 2 1000 ¸ ∙ 1 1 2πx400 ωT p = sin 2 ) = 0. Thus pk = qejφ k .4 Consider the specifications of the low pass filter in Example 10. k = 0. The roots in the p-plane occur on a circle of radius q. . (n − 1) and (2k + 1)π for n even n 2kπ φk = for n odd n φk = (10. (10.46) 207 .41) where q = sin2 (ωcT /2) and p = (z − 1)2 / − 4z. 1.42) (10. that is −4pz = (z − 1)2 = z 2 − 2z + 1 z 2 − 2z(1 − 2p) + 1 = 0 z = (1 − 2p) ± (4p(p − 1)) (10.9 045 1+ q 0.3 and design the digital filter by direct method.0 955 n (10. . Example 10.44) therefore or Hence it is seen that for every root in the p-plane.43) Having solved for p. Solution: ¸ 1 2πx100 ωc T 1 = (1 − cos ωcT ) = (1 − cos ) = 0.45). there will be two corresponding roots in the z-plane.39) |G(ejωT )|2 = where ωc is the desired angular cutoff frequency.

50) z 0 = 0.52) (10.268)] o1/2 π n π π ± 4(.417 + j0.09556 )[0.20 = 100 = 89 = n= 10 Log10 [1 + (9.09556 3π 2 (10.53) Now taking |G(j ω)| = 1 at ω = 0.4117 208 (10.09556 )[(0.268)]| G(z) = 0. and φ0 = and φ 1 = 3π 2 Therefore P0 = 0.583 − j0.09556 4(0.56) Hence .49) (10.246 [(z − (0.268)][(z − (0.45).459) Therefore z1 = 0.649 (outside the uni-circle) Similarly ¸½ ¾1/2 ∙ 3π 3π 3π zi = 1 − 0.159)(10.09556 Therefore and z 0 = 1.583 + j0.09556 Now applying Eq. z 0 = 0.583 + j0.166z + 0.649 (outside the unit circle) For stability we use the poles which are inside the unit circle.47) (10.47)n 2 (10.583 − j 0.50) (10.55) (10.268 (inside the unit circle) z1 = 1.191) ± (0.09556 − 1] 2 2 2 = (1 + j0. we will choose a second order digital filter.47)n ] (9.583 − j0.54) (10.246 = 2 z − 1.268)][z − (0.268)][1 − (0.583 + j 0.09556 − 1]1/2 2 2 2 = (1 − j0.583 + j0.47)n ] 1 + (9.417 − j0. therefore we obtain G(z) = 1 [z − (0. Therefore |G(1)| = |1 [1 − (0.583 − j0.268 (inside the unit-circle) π 2 andP1 = 0.191) ± (0.417 + j0. (10.268)] 0.48) π 2 Hence. then z = ejωT = 1.51) (10. Therefore k = 0 and k = 1.417 + j0.

10.246 = (cos 2ωT − 1.57) (10.2 Frequency transformation used with Direct Design Methods. ω1 and ω2 are the lower and upper cutoff frequencies respectively and T is the sampling period.166 cos ωT − j1.166ejωT 0.58) The frequency for Eq. note that in using Table 10. Also note that it is possible to transform from low pass to low pass. (10. band pass and stop filters. that is a shift of cutoff frequency.4117 (10.66 sin ωt) 0.58) can now be plotted for T = 1 msec and is shown in Fig.166 sin ωT + 0.246 = cos 2ωT + j sin 2ωT − 1.2 β(rad/sec) is the desired cutoff frequency.The frequency response of this filter can be obtained by substituting z = ejωT . Furthermore.1 may be used to frequency transform from low pass filters to high pass. 10.246 ej2ωT − 1. Table 10.477)j(sin 2ωt − 1.4117 0.166 cos ωT + 0. Thus G(ejωT ) = 0. Filter Lowpass Highpass Substitute for z 1 − az z −a i h − 1 + az z+a Design Formulae a = sin(β − ωc )T /2 sin(β + ωc )T /2 a= a= b= cos(β − ωc)T/2 cos(β + ωc)T /2 cos(ω2 + ω1 )T /2 cos(ω2 − ω1 )T /2 Bandpass z 2 (b − 1) 1 − 2abz + (b + 1) (b + 1) h i b − 1 − 2abz + z 2 b+1 (b + 1) z 2 (1 − b) 1− cos(ω2 − ω1 )T /2 1 tan βT /2 a = Same as above b= tan(ω2 − ω1 )T /ω 1 tanβT /2 Bandstop h 2az (b + 1) 1 − b − 2az +z 2 b + 1 (b + 1) i 209 . Table 10.

.

(a) Find the magnitude squared function of the G(z) = 1 + z −1 1 + 0. (a) Sketch the root loci in the z-plane as a function of k. An error-sampled control system has the block diagram shown in Fig. (i) The 3dB cutoff point is at θc = 0. design a digital compensator.1.4. Find the corresponding digital transfer function by bilinear transformation method. (ii) The 10dB attenuation period. (c) When k is set at the marginal value for stability. 10p-1. (b) What is the marginal value of k for stability. 10p-1.707.PROBLEMS 10. 10. The transfer function of the controlled process is G1 (s) = The sampling period is 1 sec. Also.5z −2 k(s + 1) s(s + 2)(s + 3) (b) Construct a pole-zero diagram of G(z). A sampled data feedback control system with digital compensation is as shown in Fig.5 sec.3. 10.2.5z − 1 + 0. 10. The transfer function of the controlled process is G1 (s) = K s(s + 1)(s + 2) The sampling period is 0.5 sec. find the pole-zero diagram of the resultant transfer function and sketch the magnitude characteristics. 211 . (a) Sketch the root loci in the z-plane as a function of k (b) What is the marginal value of k for stability. Consider the transfer function of an analog filter. so that the damping ratio of the closed loop control poles is equal to 0. design a digital compensator so that the damping ratio of the closed loop poles is equal to 0. G(s) = 1 √ 2s + 1 s2 + Let the sampling period be 0. (c) Construct the pole-zero diagram of magnitude squared function of G(z).1 π rad.5.65. 10. (c) When k is set at the marginal value for stability. Suppose that a low-pass Butterworth filter is desired to satisfy the following requirements.

6 A lowpass digital filter is required to have 3dB attenuation at 2kHz and at least 20dB attenuation at 5kHz. derive G(z) to satisfy the above specifications. Sketch the magnitude vs. Take sampling frequency as 20kHz. Using the direct approach of squard magnitude functions. 10. Find (a) The order of the Butterworth low pass filter. (c) The magnitude vs. (b) The digital equivalent filter by bilinear transformation. frequency plot. frequency plot.(iii) The sampling period T is 10π sec. 212 . (note that ωc = θc /T and ω = θ/T ).

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