2. Consider a sampler which samples the continuous-time input signal x(t) at a sampling frequency F, = 7Hz and produces at its output a sampled discrete-time signal x,(t) = x(nT,), where T, = 1/F, is thesampling period. The sampled signal is applied to a unity-gain lowpass filter with cutoff frequency ofF,/2 to get the signal y(t). Sketch the spectrum (between -2F, and 2F,) of the signals x(t), x,(t) and y(t)and determine the frequency of the signal y(t) for each of the following input signals:(a) x(t) = cos(4t t).(b) x(t) = cos(8at).(c) x(t) = cos(10nt).

Answered: 2. Consider a sampler which samples the…

Introductory Circuit Analysis (13th Edition)

13th Edition

ISBN:9780133923605

Author:Robert L. Boylestad

Publisher:Robert L. Boylestad

Chapter1: Introduction

Section: Chapter Questions

Problem 1P: Visit your local library (at school or home) and describe the extent to which it provides literature...

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Similar questions

A continuous time signal is given by: x(t) = A cos(8000t) cos (2000nt) %3D The signal is sampled with a sample period of 3 x 10-4 seconds. Can we recover the signal from the sampled version using an appropriate low pass filter?
Draw frequency response curve of a bandpass filter whose end frequencies are 0.1 MHz and 0.4 MHz respectively. Consider peak amplitude as 1. Use your own scale and necessary values.
Consider the analog signal x(t) = 3 cos2000πt+5 sin 6000t+10cos12,000 t i. What is the Nyquist rate for this signal ii. Assume now that we sample this signal using a sampling rate S = 5000 Hz. a) Find the reconstructed signal. b) What is the discrete-time signal obtained after sampling?
We have a biomedical measurement set-up that can be modeled as a second order system with natural resonance frequency 15.9 Hz, damping coefficient 0.707 and gain 10.a. Determine the transfer function of the system.b. Determine the differential equation representing the system.c. Design the system in block diagram form using integrators, constant multipliers, and summing junction.d. Determine the step response of the system and draw it showing all important amplitude and time values on the plot.
c) An ideal filter has exactly flat passbands with constant gain, exactly flat stopbands with zero gain, and zero-width transition bands. Sketch an example of such an ideal filter and explain why the ideal filter cannot be realised practically.
An electronic stethoscope is to be designed to operate within a full frequency range of 20 Hz to 20kHz. The accepted frequency range of human heart sounds is about 20Hz to 200Hz while that of the human lung sounds is about 25Hz to 1500Hz. Design an RC active bandpass filter circuits that could be implemented in a device to process signals that are captured from the heart sounds and the lungs sounds. Assume the voltage signals from the heart sounds from the transducer is 20mV and the desired output voltage is 5V. Use the non-inverting amplifier for your design.
Consider an analog signal x(t) which contains the frequency component up to 300 Hz. x(t) is sampled and converted to a discrete signal, x(n). Plot the approximate magnitude response of X(@) with proper normalized frequency scaling for the following sampling frequencies: i) fs-200 Hz. ii) fs = 700 Hz. For the above example what would be the band of the final signal after passing through the DAC, when the sampling frequency is taken as 100 Hz.
A signal f(t) = sinc (200rt) is sampled (using uniformly spaced impulses) at a rate of (a) 150 Hz (b) 200 Hz (c) 300 Hz. For each of the three cases (i) sketch the spectrum of the sampled signal, (ii) explain if you can recover the signal f(t) from the sampled signal, (iii) if the sampled signal is passed through an ideal lowpass filter of bandwidth 100 Hz, sketch the spectrum of the output signal. %3D
2. a) Consider an analog signal x(t) which contains the frequency component 0 Hz to 300 Hz. x(t) is sampled and converted to a discrete signal, x(n). Plot the approximate magnitude response of X(e) with proper normalized frequency scaling for the following sampling frequencies: i) fs = 300 Hz.
1.BOlu.. In the system shown in Figure (a), m(t). It represents a low frequency signal with a bandwidth of w Hz. H(), which is the frequency response of the band-pass filter. It is given in Figure (b). Since the output signal is desired to be ze=10-x(+), which of the following must the maximum value of the W bandwidth (in Hz) be lower? x() - m(t)-cos(2400zr1)| () = 10x(1)+x²(1) z(1) Strengthening Band-Pass Filtered Şekil (a) H(S) S(Hz) -1700 -700 700 1700 Şckil (b) Figure (b) O 450 Hz O 550 Hz About 550 Hz O 250 Hz About 250 Hz
3. An analog signal xa(t) has frequency components up to and including 100 kHz (2 = 200 rad/s). We desire an output ya(t) to be a filtered version of the input where only frequency components between 0 and 40 kHz are preserved. The filtering is to be accomplished by sampling the cotinuous-time signal and using a digital filter. a) What is the minimum sampling rate that can be used? b) For the minimum sampling rate found in (a), what range of digital frequencies o should be rejected by the filter? c) Sketch the magnitude characteristic of an ideal digital filter that will accomplish the desired filtering operation.
need help - nyquist and sampling process

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