Sound is created by areas of high and low pressure emanating from a source, (such as a speaker), and ultimately reaching your ear-drums, which in turn send a signal to your brain (pay attention in Science class!). If you watch an audio speaker, such as a bass woofer, when music is playing you can see the paper cone moving in and out from the magnet center.  This movement pushes and pulls the air, generating waves of high and low pressure.  When these pressure cycles reach your ears your brain interprets them as sound.

In studying and working with sound waves they are usually shown on a graph, with different aspects of the wave indicating the pressure, frequency, and direction at each point in time.  Music wave forms can appear very complicated, but a simple wave form is shown as a sine wave.

The height of each node represents the amplitude, or volume, measured in Sound Pressure Level or decibels (dB).  The length of each complete wave represents the frequency. Frequency is measured in cycles per second, or Hertz, (Hz).

Fig.'s 1 to 4 show a comparison of sound waves of different amplitude and frequency.  Fig. 2 is of higher amplitude (louder) than Fig.1, but they are both the same frequency or pitch.  Imagine hitting the same key on a piano, first softly, then harder.

Fig. 3 is a lower frequency than Fig. 4, but they are both the same volume.  Imaging hitting a low key on a piano, then hitting a higher key, but with the same force on each.

In terms of music and sound, analog audio refers to the way that most sound systems operate.  From the original music source to the microphone or recording device, sound waves are converted into electrical energy, with the frequency and intensity of the electrical signal being proportionate, or analogous, to the sound waves entering the device. From there, the electrical signal is increased in voltage, (amplified), hundreds or thousands of times, then output to the speakers.  The speaker then converts the varying electrical signal back into a back-and-forth motion, which re-creates a version of the original input sound waves.

Variations in the grooves of a phonograph record, and variations in the strength of magnetic charge in audio tapes store these types of signals.  Unfortunately, the physical condition of these storage media can degrade, and significantly lower the quality of the reproduced sound.

A computer understands only zeros and ones, and not voltage signals of varying strength.  In digital audio the sound is measured, (sampled), at regular intervals and converted into a digital signal. The sample at each interval is converted into a digital number that represents the value signal.  One might think that a sound wave broken up into small digital fragments might sound choppy, but the frequency of samples is so great that our ears perceive a high quality representation. A typical sampling rate for CD quality sound is 44.1 kHz, or 44 100 times a second. 

Terms such as 8 bit and 16 bit recording refer to the resolution of a digital signal.  This is the range of values that can be assigned to each sample. CD audio uses a 16 bit depth, which provides a range of values from 0 to 65,534 (2¹⁶). 8 bit sound would have 256 levels, (2⁸). The higher the bit depth, the better the reproduction of the original sound recording.

The number of samplings per second is called Sample Rate

The number of parts the level of input signal is divided into is called Bit Depth