I want to solve this task, in addition to having comments inside the code to explain the steps, and thanks a lot .

1. Record a voice signal of your own choice and store it in an onedimensional array called myRecording. You can use the following MATLAB code to record a voice signal a sampling frequency of 8000Hz : recobj = audiorecorder; disp('Start speaking.') recordblocking (recobj, 3) ; disp('End of Recording.'); play (recobj); myRecording = getaudiodata ( recobj ); plot (myRecording) ; 2. Create a low pass filter and store it in a vector called h, as shown below: \& Filter parameters: FS=8000; sampling frequency fc =250; cutoff frequency % size of the filter is the same as the recorded signal L=size(myRecording,1); % Design the filter using the window method: x=((L1)/2:(L1)/2); hideal =sinc(2xfCcFs); h = hamming (L) ' . * hideal; h is our filter h=h; figure plot (h (L/2-300:L/2+300)) Task 1: Time-domain analysis 1. Use the convolution operation to apply the filter, h, to the signal, myRecording. The filtered signal should be stored as myRecording_CONV_Filtered. 2. Plot myRecording, h, and myRecording_CONV_Filtered. 3. Play the signal myRecording_CONV_Filtered using the following MATLAB code: player = audioplayer(myRecrding_CONV_Filtered, 8000) ; play (player) 4. Discuss the resulting voice in comparison to the original voice signal (i.e., discuss the impact of the low pass filter on the original voice signal)