The dsPIC Digital Signal Controller (DSC) G.726A Speech Encoding/Decoding Library performs toll-quality voice compression and voice decompression. The library is an implementation of the ITU-T G.726 (Annex A) standard, on the dsPIC DSC. The encoding algorithm used is Adaptive Differential Pulse Code Modulation (ADPCM). The compression can be configured to be either 3.2:1, 4:1, 5.33:1 and 8:1, corresponding to output data rates of 40, 32, 24 and 16 kbps respectively. A well-defined API makes the library easy to integrate with the application. The G.726A library is suitable for both half-duplex and full-duplex systems. This library is available for free download and there are no associated royalties for this library. Some key applications include:
• Intercoms
• Emergency phones
• Walkie-talkies
• Mobile hands-free kits
• Digital radios
• Voice-over-IP telephony
• Building and home safety systems
• Smart appliances
• Voice recorders
• Answering machines
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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, like a source file, and built into the application.
Resource Requirements: (Encoder & Decoder)
Please refer to user’s guide for resource requirements
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Key Features:
• Free download
• Fixed 8 kHz input sample rate
• User-selectable output data rate of 40, 32, 24 or 16 kbps
• PESQ-based Mean Opinion Score (MOS): 4.3 to 4.5 (out of 5.0)
• Adaptive Differential Pulse Code Modulation (ADPCM) based coding
• Playback-only applications benefit from the Speech Encoder Utility. Encoded files can be created from the desktop using a PC microphone or WAV file.
• Full compliance with Microchip's MPLAB® C Compiler for dsPIC DSCs
• dsPIC® DSC G.726A Speech Encoding/Decoding help file assists in using the library
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Devices Supported:
