Use MATLAB to read an audio file. Design a low pass FIR filter with fi= fm where fm is maximum frequency component in the spectrum. Now, design a high pass FIR filter with fn = fm. Pass the original audio signal through these filters and store the outputs in two separate variables y_lp and y_hp respectively. Now, use MATLAB built-in functions to compute Fourier Transform of y_lp and y_hp. Add the resultant transforms to obtain f_out. Transform f_out back to the time domain and obtain the following: a. Display the plot of amplitude per sample. b. Display the single-sided amplitude vs frequency spectrum plot of original signal, y_lp, and y_hp. c. Comment on the resultant signal's audio quality. d. Change the values of fi and fn such that the resultant signal is exactly equal to the original audio signal.