The PIC24/dsPIC^® Digital Signal Controller (DSC) G.711 Speech Encoding/Decoding Library performs toll-quality voice compression and voice decompression. The library is an implementation of the ITU-T G.711 standard on the dsPIC DSC. The encoding algorithm used is either A-law or μ-law companding (user-selectable), and features a 2:1 compression ratio. G.711 uses minimal computational resources, and a well-defined API makes it easy to integrate with the application.
The G.711 library can be used for both half-duplex and full-duplex systems. However, due to its high output data rate, it is most suitable for full-duplex communications applications that do not need to store the encoded speech for subsequent playback. Some target applications include:

A flexible analog interface gives your design several options to consider. The speech encoder samples speech at 8 kHz using either an external codec or the on-chip 12-bit Analog-to-Digital Converter (ADC). The speech decoder plays decoded speech through an external codec or the on-chip Pulse Width Modulator (PWM). Storing compressed speech for playback requires 8 KB of memory for each second of speech.
A PC-based Speech Encoder Utility program creates encoded speech files for playback. Encoded speech files are made from either a PC microphone or existing WAV file. Once the encoded speech files are created, they are added to an MPLAB® IDE project, just like a regular source file, and built into the application.

Resource Requirements
Please refer to user’s guide for resource requirements

Key features of the G.711 Speech Encoding/Decoding Library:

Devices Supported