Imagine using your computer to record yourself talking. First, you speak into a microphone that you have plugged into your sound card. The ADC translates the analog waves of your voice into digital data that the computer can understand. To do this, it samples, or digitizes, the sound by taking precise measurements of the wave at frequent intervals.
The number of measurements per second, called the sampling rate, is measured in kHz. The faster a card's sampling rate, the more accurate its reconstructed wave is.
If you were to play your recording back through the speakers, the DAC would perform the same basic steps in reverse. With accurate measurements and a fast sampling rate, the restored analog signal can be nearly identical to the original sound wave.
Even high sampling rates, however, cause some reduction in sound quality. The physical process of moving sound through wires can also cause distortion. Manufacturers use two measurements to describe this reduction in sound quality:
- Total Harmonic Distortion (THD), expressed as a percentage
- Signal to Noise Ratio (SNR), measured in decibels
For both THD and SNR, smaller values indicate better quality. Some cards also support digital input, allowing people to store digital recordings without converting them to an analog format.
Next, we'll look at the other components commonly found on sound cards and what they do.