Before we can start working with sound in a computer, we need to create a digital version of it. Real life sound is continous. Digital sound consists of many small snapshots. CD quality sound consists of 44100 snapshots per second, with each snapshot consisting of 16 0s or 1s, giving us a binary representation of the amplitude at that point in time. In music tech we call this the sample rate (44100 hz in this case) and bit depth (16 bits). There are many possible combinations of sample rate and bit depth, but for most uses the 44100/16 version is good enough.
There are different file formats for digital sound. They can be divided into two categories: compressed or lossless. For work with computer sound you should always work with a lossless format. The two most commonly used of these are WAVE (Microsoft) and AIFF (Apple), and you can see what they are by looking at the file extension:
recording-1.wav // Microsoft recording-2.wav // Microsoft recording-3.aif // Apple
It makes no difference which one you choose, any software can read both of them.
The most common compressed format is MP3. If you have to use this in a project you should convert it to WAV or AIFF before doing anything else. Compression of a sound file will always degrade the quality of the signal.
With severe compression at 8kbs, 24000 hz: