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Q1:5) Find the system response to the input x(t), given the impulse response given by h(t), and as described in the figures below (hint: Graphical Convolution), where: x(t) = t & h(t) = tSee Answer
Q2: An actie filter has been designed as shown. Determine the (а)Order of the low pass filter; what are the Butterworth Coefficients? (b) (3 dB) cutoff frequency (c) Approximate Stopband frequency and stopband attenuation by calculation. You can confirm your results using MultiSim See Answer
Q3: Consider a hydraulic press that lifts a weight of 100 kg by applying a 10 kg force. What is the ratio of area of the load' piston to the area of the 'force'piston? А. V10В. 10C. V0.1D. 0.1See Answer
Q4: (а)Determine the DC conditions of the amplifier (b) Draw the AC model of the amplifier (c)Determine the midband voltage gain A, (you can use Sim) (d) Determine the low frequency poles of the amplifier (You can calculate the low frequency poles, or find the pole due to Cl by making Cs and C2 equal to 1F, etc for each pole and each capacitor). If you simulate to find the low frequency poles, please include the resulting frequency responses. 3. The enhancement NMOSFET used in the amplifier shown has the following specifications: K=0.16 mA/V² , V7=2 V, and V,= 300 V. You can show Simulations results for each part – just make sure you highlight the results as a text in your simulations.See Answer
Q5: Question 1: Using two 74192 ICs decade counters, two 7447 ICs 7-segment decoders, two 7- segment displays and negative-edge triggered D-flipflop. Design a box counting system that works with a conveyor belt. The system should stop the conveyor belt when the count of boxes reaches 23 boxes. A manual push button should reset the count and restart the working process of the conveyor belt.See Answer
Q6:Consider the following signal, x(t): Draw y(t)=-2x(-1/3t+1). Label both axes on your plot completely.See Answer
Q7:Problem 3 [15 points] A commonly used method for converting discrete-time signals to continuous- time signals is zero-order hold, in which z(t) is held constant between its known samples at t = nTg. Zero-order hold interpolation of samples is illustrated in the following figure a. [10 points] Zero-order hold is a low pass filter. Compute its frequency response H(w). Hint: The sampled signal is z, (t) = a[n]d(t-nT) and the zero-order hold signal is equivalent to convolving the sample signal r(t) with a rectangular pulse of with T. b. [5 points] Zero-order hold is used to reconstruct cos(0.1t/T.) from its samples at t = nT,. The first copy of the spectrum induced by sampling lies inSee Answer
Q8:Problem 4 [20 points] Use MATLAB command tf and bode, generate magnitude (in decibel) and phase plots (in degree) for the following voltage transfer functions.See Answer
Q9:Problem 5 [10 points] A signal z(t), bandlimited to 10 Hz, is sampled at 12 samples/s. what portion of its spectrum can still be recovered from its samples? Please draw a sketch (with an arbitrary signal shape of bandwidth 10Hz) that illustrates your findings.See Answer
Q10:Problem 6 [25 points] What does aliasing sound like? Load the file P673.mat from the Canvas. This is a speech signal (a single sentence) sampled at 24000 samples/s. a. [5 points] Listen to the signal using Describe what you hear. b. [5 points] Plot the one-sided magnitude spectrum from 0 to 8 kHz using N-length (X)/3; F=linspace (0,8000,N); FX=abs(fft (X)); plot (F,FX(1:N)) c. [5 points] Repeat (a) and (b) after reducing the sampling rate to 6000 samples/s. Do this by keeping only every fourth sample and discarding the other three samples. Use Y=X(1:4:end); soundsc (Y,6000) Describe what you hear. It should sound different. d. [5 points] Plot the one-sided magnitude spectrum of the signal in (c) from 0 to 3 kHz using Nl-length (Y)/2; Fl=linspace (0,3000, N1); FY=4*abs (fft (Y)); plot (F1,FY(1:N1)) e. [5 points] Compare (note differences) answers to (a) and (c), and to (b) and (d).See Answer
Q11:Question 3 The system of the figure below is to be used to filter continuous time music signals using a sampling rate of 6 kHz. The lead singer of the group sings at the frequencies A=1.3 kHz~ B=2 kHz.See Answer
Q12:Question 4 You are asked to sample the two sinusoidal signals below:See Answer
Q13:An ideal digital differentiator is described by the system y[n] = (x[n=1]-x[n-1] is the system LTI?See Answer
Q14:Problem 4 [15 points] If input a[n] = 9+2 cos(n) +3 cos(n), and â† [u]x 5y[n]+3y[n-1]+ y[n âˆ’ 2] = 7x[n]+6x[n 1] x[nâ€”2] a. [7 points] Compute the frequency response H(e). b. [8 points] Compute the output y[n].See Answer
Q15: Given x[n] = {..., 12, 9, 6, 3, 6, 9, 12, 9, 6, 3, 6, 9, 12,...}, a. Compute its DTFS expansion. Hint: [n] has period = 6, is real, and is an even function. b. Compute its average power in the time domain. c. Compute its average power in the frequency domain. By Parseval's theorem, both should match. Note: fft in Matlab can be used to check the results, but all work needs to be shown for full credit.See Answer
Q16:2.1. Let x[n] = 8[n] + 28[n 1]-8[n-3] and h[n] = 28[n+ 1] + 28[n − 1]. Compute and plot each of the following convolutions: (a) yi[n] = x[n] *h[n] (c) y3[n] = x[n] *h[n + 2] (b) y2[n] = x[n+ 2] *h[n]/nQ.1 Textbook P.2.1 (a) & (b) modified. Compute each of the following convolutions below (modified from the textbook). Provide your answer in a mathematical description and no need to plot it. (a) y[n] = x[n]*h[n] (b) y[n] = x[n+2]*h[n-1]See Answer
Q17:2.3. Consider an input x[n] and a unit impulse response h[n] given by -2 - (1) ²² x[n] u[n - 2], h[n] u[n + 2]. Determine and plot the output y[n] = x[n] *h[n]./nQ.2 Textbook P.2.3 modified. Use h[n] = u[n-2] instead.See Answer
Q18:2.8. Determine and sketch the convolution of the following two signals: [1+1, 0≤t≤1 1 <1 ≤ 2, elsewhere x(t) = 2-1, 0, h(t) = 8(t + 2) + 28(1 + 1)./nQ.4 Textbook P.2.8See Answer
Q19:2.9. Let h(t) = e²¹u(-t + 4) + e 2¹u(t-5). Determine A and B such that h(t-T) = e-2(1-T), T<A 0, e2(1-7), A<T<B. B<T/nQ.5 Textbook P.2.9See Answer
Q20:3.2. A discrete-time periodic signal x[n] is real valued and has a fundamental period N = 5. The nonzero Fourier series coefficients for x[n] are ao = 1, a₂ = a^₂ = ej/4,a₁ = a₁ = 2e/m/3 Chap. 3 Problems k=0 Express x[n] in the form +ΣA x[n] = Ao + WI A sin(wn + x). 251See Answer

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