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to mainstream usage. In 1996, in a book on multimedia on Linux, we wrote about a technology called MPEG-1 layer 3, or MP3. At the time it was relatively unknown, used only by some obscure web sites to distribute music, and my then-current 40 MHz Intel 386 computer was barely able to decode it in real time. Not so many years later, it has become ubiquitous and the de facto standard file format for digital music on the Internet. At the same time, other technologies that appeared promising have fallen by the wayside, often not for technical reasons. To stay current, check the resources listed at the end of the chapter.

There are minor differences among Linux distributions. Although most of the information in this chapter is generic and applicable to most Linux distributions, for details you should consult the documentation that came with your system, contact your distribution vendor, or consult with fellow users.

Multimedia Concepts

This section very quickly covers some concepts relevant to digital audio , video , and sound cards . Understanding these basics will help you follow the rest of the material in this chapter.

Digital Sampling

Sound is produced when waves of varying pressure travel though a medium, usually air. It is inherently an analog phenomenon, meaning that the changes in air pressure can vary continuously over a range of values.

Modern computers are digital, meaning they operate on discrete values, essentially the binary ones and zeroes that are manipulated by the central processing unit (CPU). In order for a computer to manipulate sound, then, it needs to convert the analog sound information into digital format.

A hardware device called an analog-to-digital converter converts analog signals, such as the continuously varying electrical signals from a microphone, to digital format that can be manipulated by a computer. Similarly, a digital-to-analog converter converts digital values into analog form so they can be sent to an analog output device such as a speaker. Sound cards typically contain several analog-to-digital and digital-to-analog converters .

The process of converting analog signals to digital form consists of taking measurements, or samples, of the values at regular periods of time, and storing these samples as numbers. The process of analog-to-digital conversion is not perfect, however, and introduces some loss or distortion. Two important factors that affect how accurately the analog signal is represented in digital form are the sample size and sampling rate.

The sample size is the range of values of numbers that is used to represent the digital samples, usually expressed in bits. For example, an 8-bit sample converts the analog sound values into one of 28, or 256, discrete values. A 16-bit sample size represents the sound using 216, or 65,536, different values. A larger sample size allows the sound to be represented more accurately, reducing the sampling error that occurs when the analog signal is represented as discrete values. The trade-off with using a larger sample size is that the samples require more storage (and the hardware is typically more complex and therefore expensive).

The sample rate is the speed at which the analog signals are periodically measured over time. It is properly expressed as samples per second, although sometimes informally but less accurately expressed in Hertz (Hz) . A lower sample rate will lose more information about the original analog signal, a higher sample rate will more accurately represent it. The sampling theorem states that to accurately represent an analog signal it must be sampled at at least twice the rate of the highest frequency present in the original signal.

The range of human hearing is from approximately 20 to 20,000 Hz under ideal situations. To accurately represent sound for human listening, then, a sample rate of twice 20,000 Hz should be adequate. CD player technology uses 44,100 samples per second, which is in agreement with this simple calculation. Human speech has little information above 4000 Hz. Digital telephone