Upgrading and Repairing PCs

Audio Adapter Concepts and Terms

To fully understand audio adapters and their functions, you need to understand various concepts and terms. Terms such as 16-bit, CD quality, and MIDI port are just a few. Concepts such as sampling and digital-to-audio conversion (DAC) are often sprinkled throughout stories about new sound products. You've already learned about some of these terms and concepts; the following sections describe many others.

The Nature of Sound

To understand an audio adapter, you must understand the nature of sound. Every sound is produced by vibrations that compress air or other substances. These sound waves travel in all directions, expanding in balloon-like fashion from the source of the sound. When these waves reach your ear, they cause vibrations that you perceive as sound.

Two of the basic properties of any sound are its pitch and intensity.

Pitch is the rate at which vibrations are produced. It is measured in the number of hertz (Hz), or cycles per second. One cycle is a complete vibration back and forth. The number of Hz is the frequency of the tone; the higher the frequency, the higher the pitch.

Humans can't hear all possible frequencies. Very few people can hear sounds with frequencies less than 16Hz or greater than about 20KHz (kilohertz; 1KHz equals 1,000Hz). In fact, the lowest note on a piano has a frequency of 27Hz, and the highest note has a frequency a little higher than 4KHz. Frequency-modulation (FM) radio stations can broadcast notes with frequencies as high as 15KHz.

The amazing compression ratios possible with MP3 files, compared to regular CD-quality WAV files, is due in part to the discarding of sound frequencies that are higher or lower than normal hearing range during the ripping process.

The intensity of a sound is called its amplitude. This intensity determines the sound's volume and depends on the strength of the vibrations producing the sound. A piano string, for example, vibrates gently when the key is struck softly. The string swings back and forth in a narrow arc, and the tone it sends out is soft. If the key is struck more forcefully, however, the string swings back and forth in a wider arc, producing a greater amplitude and a greater volume. The loudness of sounds is measured in decibels (db). The rustle of leaves is rated at 20db, average street noise at 70db, and nearby thunder at 120db.

Evaluating the Quality of Your Audio Adapter

The quality of an audio adapter is often measured by three criteria: frequency response (or range), total harmonic distortion, and signal-to-noise ratio.

The frequency response of an audio adapter is the range in which an audio system can record or play at a constant and audible amplitude level. Many cards support 30Hz–20KHz. The wider the spread, the better the adapter.

The total harmonic distortion measures an audio adapter's linearity and the straightness of a frequency response curve. In layman's terms, the harmonic distortion is a measure of accurate sound reproduction. Any nonlinear elements cause distortion in the form of harmonics. The smaller the percentage of distortion, the better. This harmonic distortion factor might make the difference between cards that use the same audio chipset. Cards with cheaper components might have greater distortion, making them produce poorer-quality sound.

The signal-to-noise ratio (S/N or SNR) measures the strength of the sound signal relative to background noise (hiss). The higher the number (measured in decibels), the better the sound quality. For example, the top-of-the-line Sound Blaster Audigy 2 sound card features an SNR of 106db, whereas the older Sound Blaster Audigy is rated at 100db and the AWE64 series has an SNR of 90db.

These factors affect all types of audio adapter use, from WAV file playback to speech recognition. Keep in mind that low-quality microphones and speakers can degrade the performance of a high-quality sound card.

Sampling

With an audio adapter, a PC can record waveform audio. Waveform audio (also known as sampled or digitized sound) uses the PC as a recording device (like a tape recorder). Small computer chips built into the adapter, called analog-to-digital converters (ADCs), convert analog sound waves into digital bits that the computer can understand. Likewise, digital-to-analog converters (DACs) convert the recorded sounds to an audible analog format.

Sampling is the process of turning the original analog sound waves into digital (binary) signals that the computer can save and later replay (see Figure 16.7). The system samples the sound by taking snapshots of its frequency and amplitude at regular intervals. For example, at time X the sound might be measured with an amplitude of Y. The higher (or more frequent) the sample rate, the more accurately the digital sound replicates its real-life source and the larger the amount of disk space needed to store it.

Originally, sound cards used 8-bit digital sampling that provided for only 256 values (2⁸), which could be used to convert a sound. More recently, sound cards have increased the quality of digitized sound by using 16-bit (2¹⁶) sampling to produce 65,536 distinct values. Today's highest-quality sound cards feature 24-bit sampling (2²⁴), which translates into more than 16.8 million possible digital values that can be matched to a given sound.

Note For more information on the differences between 8-bit and 16-bit audio sampling, see "8-Bit Versus 16-Bit" in Chapter 16 of Upgrading and Repairing PCs, 13th Edition, included on the DVD-ROM accompanying this book.

You can experiment with the effects of various sampling rates (and compression technologies) by recording sound with the Windows Sound Recorder or a third-party application set to CD-quality sound. Save the sound and play it back at that highest quality setting. Then convert the file to a lower-quality setting, and save the sound file again with a different name. Play back the various versions, and determine the lowest quality (and smallest file size) you can use without serious degradation to sound quality.