Question

in the real world humans hear sound as an analog signal. this means the signal is a continuous waveform which is completely processable by human ears. however machines 'hear' sound a little differently as they process sound digitally which is a discrete waveform. when sound is recorded or transmitted electronically the continuous (analog) waveform is sampled to convert it to a discrete (digital) sequence. sampling is the process of reducing a continuous time signal to a discrete-time signal. as the sampling rate increases the sound quality of the recording or transmission will improve. the graphs below represent two different samples of pure tone. sample one is taken 8 times per unit of time. sample two is taken 16 times per unit of time. a) why would sample 2 reproduce a better replication of the pure signal? b) write a sinusoid model for sample 2

Answer 1

Answer for A since the other person answered only B

Sample 2 would reproduce a better replication of the pure signal because it records more frequently, meaning that it would be closer to being a continuous waveform when compared to sample 1.

Answer 2

y=7.75cos(pi/8(x)) + 7.75

Explanation:

A= 15.5 - 0 = 15.5/2 = 7.75 (Max Y - Min Y)

P = 16-0 (Distance between Max x and next Max x)

F= Pi/8

Midline= 7.75

Vertical Shift = Midline = 7.75