This is in electrical engineering signals, I NEED IT ASAP PLEASE!!
1) Given input signal is x(t), and output y(t). The frequency response of the system is H(jomega).
If H(jomega) has impulse response of h(t), the relation between the input and output signals is given by using the convolution in time domain as:
Convolution in time domain will correspond to multiplication in frequency domain, Hence, the relation in frequency domain is:
The above gives the relation of fourier transform of input and output signals.
2) given X(jomega)
and H(jomega)
By using the relation:
The fourier transform of output signal is the multiplication of fourier transform of the input and the system transfer function:
H(jomega) is zero in range 2pi100 and 2pi50, in that range the output fourier transform is zero.
Hence Y(jomega) =
c) Note that any sampled signal can be reconstructed only if it satisfies sampling theorem.
The sampling theroem states that the sampling frequency should be greater than or equal to twice the message frequency or message signal band width.
Fb is the bandwidth of message signal
From the fourier transform of the input signal X(jomega), observe that the band width of the input signal is
200, since w = 2*pi*f, from X(jomega) f lies between +100 and -100.
Hence the sampling frequency should be,
Given Fs = 400,
Given sampling frequency is 400, which satisfies the sampling theorem ( signal has minimum sampling frequency).
To match the output signal the filter band width also should be reduced by same amount. The band width of given digital filter is
pi/2+pi/2 = pi
But the analog filter has bandwidth of
100pi+100pi = 200pi
The digital filter bandwidth is
To acheive this band width the cutoff frequency of filter should be
pi/4, hence
pi/4+pi/4 = pi/2
By changing the filter, cutoff frequency, we can reconstruct the output signal.
This is in electrical engineering signals, I NEED IT ASAP PLEASE!! (b) A continuous-time LTT system...
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