Originally posted by Shotgun:
Let me take a shot.
If I am guessing correctly, it should be like this.
Audio signals are analogue. Means the waves are curvy. However, to convert files to mp3s, the files need to be in digital form. So the analogue wave needs to be converted to digital.
Imagine a simple positive curve in an analogue wave. Now to turn that wave into a digital wave, it becomes a square block if u use a single bit to represent it. Of course, if u use just 1 bit, u lose all the sound quality there is to it. So the idea is to use many bits to match the single curve as well as possible.
The more bits, the better the quality.
Shotgun de shot so zun eh.
http://en.wikipedia.org/wiki/Sampling_rateTS, next time must remember to use Wiki and Google first hor.
Anyway, to add on, just think of sampling rate like the frame rate of a video. Since videos are basically many images displayed successively, as you've already know, if a video has a higher frame rate (more images per second), it will appear more smooth, and hence a better quality of the video is achieved. The same goes for the sampling rate of a sound/music.
