3.20  Exercises


3.1. When the input x1[n] = δ[n] + δ[n 1] is applied to a linear system, the observed output is y1[n] = 4δ[n 1] + δ[n 2]. Inform the expression of the output y2[n] when the input is x2[n] = 3δ[n] + 3δ[n 1].
3.2. When the input x1[n] = δ[n] is applied to a LTI system, the output is y1[n] = 3δ[n] + 2δ[n 1] + δ[n 2]. What is the output y2[n] when the input is x2[n] = 5δ[n] + 2δ[n 1]?
3.3. When the input x1[n] = δ[n] is applied to a system, the output is y1[n] = 3δ[n] + 2δ[n 1] + δ[n 2], and when the input is x2[n] = 2δ[n 1], the output is y2[n] = 6δ[n 1] + 2δ[n 2] + 2δ[n 3]. Can this system be LTI?
3.4. A time-invariant system has output y1(t) = δ(t 1) δ(t 4) when the input is x1(t) = δ(t). Inform the expression of the output y2(t) when the input is x2(t) = δ(t 3).
3.5. The input-output relations that describe three given systems are: 1) y[n] = C(x[n])2, where C = 2e2, 2) y[n] = (n 1)x[n] and 3) y[n] = 3x[n 1] + 2πx[n 3]. Classify whether or not these systems are: a) causal, b) linear, c) memoryless, d) stable in BIBO sense and e) time-invariant.
3.6. A discrete-time LTI is characterized by the difference equation y[n + 2] = 5x[n + 2] + 2x[n + n0], where x[n] and y[n] are the input and output signals, respectively. Inform a value for n0 that makes the system causal.
3.7. What is the convolution matrix H for an impulse response h[n] = 3δ[n] + 2δ[n 1] + δ[n 2] when the input signals are: a) column vectors of dimension 3, b) row vectors of dimension 4?
3.8. Four continuous-time signals: x1(t) = ej4t, x2(t) = ej4t, x3(t) = cos (4t) and x4(t) = e(1+j4)t, are individually transmitted over a causal system H(s) = 1 s+10. What are the respective outputs y1(t),,y4(t)?
3.9. Four discrete-time signals: x1[n] = 5ej2n, x2[n] = ej2n, x3[n] = cos (2n) and x4[n] = (3ej2) n, are individually transmitted over the causal system H(z) = 1 10.6z1. What are the respective outputs y1[n],,y4[n]?
3.10. What is the result of the periodic convolution between a pulse train x(t) with pulses with duty cycle of 2 s, amplitude of 5 V and a period of 10 s with itself? Assume x(t) is an even signal.
3.11. The following commands were used to calculate y1 and y2:

1N=3;M=4;x=1:M;h=[0.4,-0.1];y1=conv(x,h);y2=ifft(fft(x,N).*fft(h,N));

a) What is the minimum value of N to make y1 and y2 coincide? b) Generalize the previous answer for an arbitrary value of M and length(h).
3.12. A band-limited signal x(t) was sampled obeying the sampling theorem with Fs = 10 Hz and the result is xs(t) = 4δ(t) + 3δ(t 0.1) 2δ(t 0.4). Provide the exact expression of x(t). Hint: this expression is given in terms of scaled and shifted sincs.
3.13. An AM radio signal x(t) has a bandwidth of 10 kHz at baseband and is centered at fc = 600 MHz (the passband signal occupies 20 kHz). Use undersampling to obtain a discrete-time baseband version xb[n] of this signal via digital signal processing based on a signal x[n] obtained with the minimum sampling frequency that you think is possible assuming realistic (non-ideal) filters. Describe the whole scheme informing the adopted: a) analog anti-aliasing filter, b) sampling frequency, c) any required digital filtering and/or frequency digital down-conversion, and d) phase inversion compensation (in case the signal replica at positive frequency fc was moved to a lower frequency than the replica at fc).
3.14. A band-limited even and real-valued discrete-time signal x[n] has the spectrum X(ejΩ) described in Figure 3.86 with Ω1 = π4 and Ω2 = π2 rad. It is converted to a continuous-time signal x(t) using the scheme depicted in Block (3.25) with Fs = 1 kHz and a ZOH modeled by h(t) = u(t) u(t Ts). a) Carefully draw |X(f)|, where X(f) = F{x(t)} in the range 3 to 3 kHz. b) Calculate X(f) (magnitude and phase) at f = 0,100 and 1200 Hz. c) Repeat the calculation in b) but now considering that x[n] is a critically sampled signal with Ω1 = π4 but Ω2 = π rad.

PIC
Figure 3.86: Spectrum of a band-limited even and real-valued signal x[n].

3.15. Assume the DUT in the circuit of Figure A.34 is the SAW filter described in Table 3.4 (assume its typical parameters). Both the source ZS and load ZL impedances are 100 Ω. a) If the input signal x(t) (denoted as V s in Figure A.34) is a sinusoid of frequency 70 MHz and power 4 dBm, what is the power over the load and a reasonable expression for the output signal y(t)? b) If x(t) = 3cos (ω1t) + 6cos (ω2t) with ω1 = 2π × 70 and ω2 = 2π × 76.5 Mrad/s, what is a reasonable expression for the output y(t)?
3.16. If the input to the filter characterized by Table 3.5 is x(t) = 4cos (2πf1t + π4) + 8cos (2πf2t), where f1 = 455 and f2 = 500 kHz, what is a reasonable expression for the output y(t)?
3.17. A non-causal analog lowpass filter has an ideal magnitude with unitary gain for frequencies 5 to 5 rad/s and zero otherwise. The phase of this filter is linear and given by ej8ω. What is the output of this filter when the input is x(t) = k=00.5k sin (1.5kt)?
3.18. A FIR filter with linear phase has an impulse response h[n] = 3δ[n] 5δ[n 1] + 7δ[n 2] 5δ[n 3] + 3δ[n 4]. Find its system function H(z) and draw a realization of this filter with the minimum number of multipliers.
3.19. a) Design two filters H1(s) and H2(s), both described by (ωn2)(s2 + 2ζωns + ωn2), with damping factors ζ = 0.707 and 0.9, respectively. In the case of H2(s), assume a natural frequency ωn = 1600 rad/s and a tolerance 𝜖 = 0.05. In both cases, the settling time must be approximately 3 ms. b) Plot and compare the graphs of the output waveforms of both systems to the step function u(t).
3.20. For the filter H(z) obtained with [b, a]=butter(3,1/3): a) find the values of H(z) for z = 0.5 + 0.5j,3 + 2j,0,1,3 and 0.5 + 0.5j. b) Using freqz.m, find the maximum value of the magnitude response and in which frequency this maximum occurs. c) What are the first five samples of the impulse response h[n]? d) Is this a FIR or IIR? e) Is this a lowpass or highpass?
3.21. For the filter H(s) = 100 s2+102s+100 (obtained with [Bs,As]=butter(2,10,’s’)); a) what are the values in Hertz of its natural and cutoff frequencies? Now consider that H(s) is the anti-aliasing filter of a DSP system operating with a sampling frequency of Fs = 4 Hz and there is a strong interferer tone with 1 dBm at 5 Hz that contaminates the signal before the filter H(s). b) At what frequency and power this interferer will appear within the band [0,Fs2[? c) And what if the interferer has the same power of 1 dBm but frequency of 50 Hz?
3.22. a) Using syntax and structure similar to Listing 3.12, list the algorithm of a digital filter implementing the system function H(z) provided by [Bz,Az]=butter(3,0.2). b) When H(z) is used in the canonical system of Figure 3.30 with Fs = 44.1 kHz and ideal analog filters, what is the corresponding cutoff frequency in Hertz of the overall system?
3.23. The filter H(s) = 1 s3+2s2+2s+1 was obtained with the command [b, a]=butter(3,1,’s’) and has cutoff frequency of 1 rad/s. Obtain a filter G(s) that corresponds to a scaled version of H(s) with a new cutoff frequency of 1200π rad/s such that both have the same gain at DC. a) Inform the linear gains |H(ω)| and |G(ω)| for both filters at frequencies ω = 0,1,1200π and 2000 rad/s. b) What are these gains in dB?
3.24. A filter H(s) = 125 s3+10s2+50s+125 has a cutoff frequency of 5 rad/s. Obtain a filter G(s) that corresponds to a version of H(s) scaled in frequency with a cutoff frequency of 3 rad/s. G(s) should also have a gain of 3 dB at DC.
3.25. a) Use the six methods of Table 3.6 to convert H(s) = 0.13(s2 0.4s + 0.13) into H(z) assuming Ts = 0.1 s and compare their frequency responses with the original one (in continuous-time) b) Repeat the procedure for Ts = 1 s. c) Discuss what methods were most impacted by the increase of Ts.
3.26. Given a filter H(s) = (0.1s2 + 71)(s2 + 11s + 71) and assuming Fs = 50 Hz, a) find H(z) using the bilinear transform, b) compare the frequency responses of H(s) and H(z) after D/C conversion using the same abscissa in rad/s, c) can you find a frequency ωm > 0 in rad/s for which the values of H(s) and H(z) are the same?
3.27. A filter H(s) = (0.1s2 + 71)(s2 + 11s + 71) should be converted to H(z) using the bilinear. The value of H(s) at s1 = j8.24 and s2 = j5 are g1 0.708ej1.5366 and g2 0.9554ej0.8743, respectively. a) Find H1(z) and H2(z) that when implemented in a hardware using Fs = 50 Hz leads to the same values g1 and g2 at ωd = 8.24 and 5 rad/s, respectively, which correspond to the frequencies Ω = ωdFs = 0.1648 and 0.1 rad. b) For comparison, show the values of Hi(ej0.1648) and Hi(ej0.1), for i = 1 and 2. c) To note the degrees of freedom, use frequency scaling such that G(s)|s=s2 = H(s)|s=s1 = g1 and then find G(z) such that when implemented in a hardware using Fs = 50 Hz leads to the same value g1 at ωd = 5 rad/s.
3.28. The filter Hs(s) = (0.1s2 + 71)(s2 + 11s + 71) should be converted to Hz(z) using the bilinear such that the value of Hs() at ω = 10 rad/s is the same value Hz(ejΩ) will have at Ω = π4 rad. a) find the value of Fs that allows to directly convert Hs(s) into Hz(z) without the traditional pre-warping step, b) assuming an arbitrary value of F s = 0.5 Hz, find the pre-warped version of Hs(s) that will generate the correct Hz(z) via bilinear with F s.
3.29. A product adopted an analog filter H(s) = 125×106 s3+103s2+500×103s+125×106, which was designed with [B,A]=butter(3,500,’s’) to have a cutoff frequency of 500 rad/s. This filter must be substituted by an equivalent system He(s) that relies on a digital filter H(z). The anti-aliasing and reconstruction filters are ideal. The ADC and DAC converters should operate at Fs = 800 Hz. Design a digital filter H(z) with the bilinear transform, such that He(s) has the same cutoff frequency of H(s).
3.30. Design a causal FIR H(z) using the windowing method. The filter must have 5 coefficients, cutoff frequency π6 rad and the adopted window must be Hann’s. The gain at DC must be unitary. The command hanning(5) in Matlab/Octave, returns [0.25, 0.75, 1.00, 0.75, 0.25].
3.31. A filter was obtained with [b, a]=butter(3,1/3). a) Draw the block diagram representation for realizations using direct form I, direct form II, transposed direct form II, cascade of SOS and parallel using SOS. b) Using any programming language, implement the direct form II. Assume that there is a function called readAD() that reads a sample from the ADC and another writeDA() that writes a sample to the DAC, and that the processing loop is correctly invoked according to the chosen sampling frequency. In other words, complete the following code (this example is in C):
1main( ) { 
2  int x, y; 
3  ... 
4  while (1) { //eternal loop 
5    x = readAD( ); 
6    ... 
7    writeDA(y); 
8  } 
9  ... 
10}