Question

In the real-world, humans hear sound as an analog signal. This means the signal

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 a pure tone. Sample 1 is taken 8 times per unit of
time. Sample 2 is taken 16 times per unit of time.

Answer 1

Answer: I hope i can help you guys unlike the other guy

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.

Explanation:

Sample two has more double more the points compared to sample 1. Sample is closer to being a continuous wave form instead of a discrete waveform when compared to sample 1. The more points a graph has, the more accurate.