Get the ex1_3.wav file and save it in C:temp\. Change the current directory in Matlab to C: temp\. This WAV-file contains a speech signal. a. Load the exl_3.wav file into Matlab. You can either use Matlab's Import Wizard, by double-clicking on the filename in Matlab's current directory window, or use the wavread command (see help wavread for details). You will need to load both the signal's samples, as well as its sampling rate b. Play the speech file in Matlab, using the soundsc command. If you've stored the signal's samples in a vector called y, and its sampling frequency in a variable called fs vou would use the command >soundsc (y,fs) Convolve the speech signal with the impulse response h[n]-f-1,2.-13. Don't worry about the index vectors here. Store the result in a new variable, say yl. Listen to the output signal using the command c. >soundsc (yl,fs) Do you hear any difference between the original signal and this filtered signal? Would you agree that this system has suppressed the lower frequencies of the signal? d. Convolve the speech signal with the impulse response h[n]-1,1,13. Again, don't worry about the index vectors. Store the result in a new variable, say y2. Listen to the output signal using the command >>soundsc (y2, fs) Do you hear any difference between the original signal and this filtered signal? What kind of system is this? Get the ex1_3.wav file and save it in C:temp\. Change the current directory in Matlab to C: temp\. This WAV-file contains a speech signal. a. Load the exl_3.wav file into Matlab. You can either use Matlab's Import Wizard, by double-clicking on the filename in Matlab's current directory window, or use the wavread command (see help wavread for details). You will need to load both the signal's samples, as well as its sampling rate b. Play the speech file in Matlab, using the soundsc command. If you've stored the signal's samples in a vector called y, and its sampling frequency in a variable called fs vou would use the command >soundsc (y,fs) Convolve the speech signal with the impulse response h[n]-f-1,2.-13. Don't worry about the index vectors here. Store the result in a new variable, say yl. Listen to the output signal using the command c. >soundsc (yl,fs) Do you hear any difference between the original signal and this filtered signal? Would you agree that this system has suppressed the lower frequencies of the signal? d. Convolve the speech signal with the impulse response h[n]-1,1,13. Again, don't worry about the index vectors. Store the result in a new variable, say y2. Listen to the output signal using the command >>soundsc (y2, fs) Do you hear any difference between the original signal and this filtered signal? What kind of system is this