Audio Hardware - Upgrading and Repairing PCs - page 843

Evaluating the Quality of Your Audio Hardware
The quality of audio hardware is often measured by three criteria: frequency response (or
range), total harmonic distortion, and signal-to-noise ratio.
The frequency response ofa soundcard orintegrated audio is the range in which an audio
system can record or play at a constant and audible amplitude level. Many cards and in-
tegrated solutions 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.
These factors affect all types of audio adapter use, from WAV file playback to speech re-
cognition. Keep in mind that low-quality microphones and speakers can degrade the per-
formance 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 (such as 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 13.1 ). The system samples
the sound by taking snapshots of its frequency and amplitude at regular intervals. For ex-
ample, 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.
Figure 13.1 Sampling turns a changing sound wave into measurable digital values.