Digital Audio Basics

Sampling rate is the number of samples of a signal that need to be captured or played back within a second to ensure the signal has been captured intelligently. The sampling rate should satisfy Nyquist criterion to prevent effects of aliasing. Nyquist criterion requires the sampling rate to be greater than twice the highest frequency in the band of interest.

Bit rate is the product of number of bits used to record a given sample and the sampling rate. It is expressed in kilobits per second (kbps).

Audio quality is directly proportional to the bit rate. However, since it can take significant memory to record an audio signal in its original form or significant time to transmit an audio signal in its original form, several software codecs such as ADPCM, G.711, G.726A and Speex have been created to compress the raw audio signal without affecting audio quality in many cases. Here’s a simple comparison:

• One second of a speech signal recorded at 16-bit resolution at 8 kHz sampling rate requires about 128 kilobits or 16 KB of memory for storage
• One second of the speech signal recorded at 16-bit resolution at 8 kHz sampling rate and encoded using the Speex library for dsPIC^® DSCs will require 8 kilobits or 1 KB for storage, providing a 16:1 compression

See the Playback and Recording sections for further details on various algorithms and on-chip peripherals used in audio applications.