Digital Audio on a Windows PC

Most PC based hardware uses a technique called Pulse Code Modulation (PCM) to store music data.  Electrical signals in the form of waves (analog signals), such as those from a microphone, are turned into numbers by a circuit called an Analog to Digital Converter (ADC) and then saved in computer memory.  The numbers that create any given sound, or music, can be saved in a file on the computers hard drive and then be manipulated very easily.  When a file is played, a Digital to Analog Converter (DAC) converts the numbers back into analog electrical signals. Audio equipment then amplifies these signals and sends them to speakers or headphones, where you hear them as sound.

The audio file format most commonly used by Windows is called a “wave” file, identified by the WAV file extension.  However, many other common formats exist.  One format is not necessarily "better" than another format.  To be more specific - you must look at the standard characteristics that they share.

These characteristics are SAMPLING RATE, BIT LENGTH , NUMBER OF CHANNELS, and COMPRESSION.

• SAMPLING RATE

When recording a sound, the ADC must select and measure a sample (a discrete instant in the sound wave), and store its amplitude (a measure of its loudness) as a number in your computer. To capture a sustained interval of sound, this process must be repeated very rapidly. The sample rate is the frequency with which the sampling process occurs.
Sample rates are generally expressed in Kilohertz (kHz), or thousands of cycles per second.  What is commonly referred to as “CD quality” requires a sampling rate of 44.1 kHz.  Since most FM Radio stations trim the size of their signal, “FM radio quality” music is typically generated at 22.05 kHz.
 

• BIT LENGTH

The amplitude of each sample is expressed as a number and, like all values stored in a computer, these are binary numbers. The electronic “digits” that represent numbers are called bits. Bit resolution (also called “bit length” or “sample size") is the number of binary digits that make up each sample. Eight bits can represent a range of values from 0 to 255, while sixteen bits can represent values from 0 to 65,535.  The larger bit length allows audio to be reproduced at a higher quality level.
 

• NUMBER OF CHANNELS (MONO OR STEREO)

A Stereo signal consists of two streams of data working together, one assigned to the left channel, the other to the right. Therefore, at any given sample rate and bit length, a stereo file requires twice as much data as its mono equivalent. A Mono signal is one stream of data that is reproduced equally on both speaker channels.  With the benefit of using only one half the required storage space, you lose stereo separation of the two audio channels.
 

• COMPRESSION

To increase the speed of processing of large amounts of data, or to shrink the hard disk space necessary to store large audio files different types of compression may be used.  Before the data is stored, some of it is removed and, on playback, this data is artificially restored.  Depending on the compression scheme used, and the original quality of the audio file before compression, the music played back can sound slightly (or greatly) different than the original sound.  Click here for more information on Audio Compression.

Table of sizes for an uncompressed 16 bit, 44.1 kHz PCM WAV music file:

The following Adobe Acrobat file compares various compression formats and their space requirements:

Digital Audio Compression Table (GRAPH.PDF)

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