Audio Signal Filtering

Background

Audio signals in the digital world are simply 1-D signals that contain the values of the sampled sound v/s an index, say k.
Consider the diaphragm on a microphone, that vibrates every time a sound impinges on it.
The vibration is converted to an electrical signal by a transducer, which then relays the "analog" signal to an A/D converter.
Finally, the A/D converter samples the analog signal, and makes it a train of samples; each box of the train contains a value.
This value corresponds to the digital representation of the electrical signal that resulted from the vibration.
For example, say the diaphragm vibrated 0.2mm, resulting in a generated voltage of 0.2mV (these values are completely arbitrary).
If the A/D converter designated 0V to x00 and 10mV to xFF, then the resolution of designating values to the samples would be 10/255 mV or .04mV. Thus, 0.2 mV would be x05.
This value would be stored in the digital sound file, against a time index corresponding to when the A/D received this sample.
Since this page focuses on Audio Signal filtering, those interested in the basics of Audio Processing can go to the references on the following wiki:

This page took a long time to figure out. Mainly because, for images and sine waves, it's easy to find some that aren't copyrighted. For images you just use your pet dog's best photograph, and for sine waves, you write a line in MATLAB. But for audio signals, almost every recorded sound is copyrighted in some way or the other, and to avoid being sued for copyright infringement, I had to find sounds to play with, and publish online, that weren't "owned", so to speak.