DSP IMPORTANT EXAMPLES

1) Compute convolution y(n)=x(n)*h(n) where x(n)={1,1,2,1,1} and h(n)={1,-2,-3,4} 2) Find the Z-transform and ROC of following sequence: (i) (ii) X1(n)= X2(n)=, ( ) ( ) ( ) ( ) . /- ( ) Z>1

3) Determine the inverse Z transform of following: X(Z)=

( )( )

4) DFT of a sequence x(n) is given by X(k)={6,0,-2,0} (i)Determine x(n) (ii) Plot x1(n) if X1(k) is X(k) (iii) Determine circular autocorrelation of x(n) using DFT and IDFT only. 5) Realise a linear phase FIR filter with the following impulse response: ( ) ( ) ) 6) Obtain the direct form II implementation of LTI system with transfer function: ( ) ( ) ( ) ( ) (

Is it advantageous than direct form I?

7) The transfer function(tf) of discrete time causal system is given by ( ) Draw the cascade and parallel realization. 8) An FIR filter is given by ( ) ( ) ( ) ( ) ( )

Find the lattice structure coefficients. 9) Find the input x(n) of the system if the impulse response h(n) and output y(n) are given by h(n) ={1,2,3,2} and y(n)={1,3,7,10,10,7,2} 10) Determine the Z transform of following sequences: (i) x(n)= for (a) a real and (b) a imaginary. Show ROC for both. (ii) x(n)= ( ) (iii) (iv) (v) (vi) (vii) ( ) ( ) ( ) ( ) 0 ( (
| |

) )

( ( )

) ( | |

( ) (

) )

. / 1 ( )

X(n)=x1(n+2) where x1(n)={1,2,3,4,5}

( ) (viii) ( ) 11) Find the inverse Z transform of following causal LTI systems: (i) (ii) (iii) ( ) ( ) ( ) | | (long division method) (PF method) () ( )

(iv) 12)

( )

( . /(

) )

| |

A causal LTI system is described by difference equation ( ) ( ) ( ) ( ) where x(n) is input and y(n) is the output. (i) Find system function H(Z). (ii) Plot zeros and poles of H(Z) and indicate ROC. (iii) Find the unit sample response of the system. (iv) Check whether the system is stable or not. Convert analog filter H(g) given by ( )
( )( )

13)

into

digital filter with sampling period T=0.5sec. Use impulse invariance method. 14) Compute linear convolution y(n)=x(n)*h(n) of the signals + + by graphical method. ( ) * ( ) * Verify the result of Z transform method. 15) Obtain DF I, II, transposed direct form II and cascade form structures of IIR system with difference equation. ( ) ( ) ( ) ( ) ( ) 16) The analog signal has a bandwidth of 4 kHz. If we use N point DFT with (m is an integer) to compute spectrum of the signal with resolution less than or equal to 25 kHz. Determine (i) minimum sampling rate. (ii) min no. of required samples N (iii) min length of analog signal record tr. 17) Compare the window functions of FIR filter design method. Determine parameters and M of Kaiser window with the following specifications. Frequency of passband edge=400 Hz. Gain in passband=-3dB Frequency with which stopband begins=800 Hz. Gain (attenuator) in stopband= -25dB.

Sampling frequency=2000Hz. 18) Design a digital butterworth low pass filter with pass band magnitude within 1dB fir frequency 0<0.2 and stopband attenuation greater than 15dB for frequency 0.3< . Use bilinear transformation method. 19) Design a LPF using Kaiser window with p=0.4, s=0.6, p=0.01, s=0.001. 20) Obtain the following structures (i) Direct forms (ii) Cascade forms (iii) Parallel forms For the system function ( ) ( . )( /( )( ) )

*note: the questions marked in red are very important.