Audio Processing Series Part IV : Encoding and Decoding audio data using ADPCM algorithm
Understanding ADPCM: Principles & Implementation ADPCM (Adaptive Differential Pulse Code Modulation) is an audio compression technique that focuses on encoding the difference between consecutive audio samples instead of their absolute values. By representing only the changes in audio data, ADPCM achieves significant data rate reductions. This project involves reading an audio file in blocks of 1024 samples, encoding it using ADPCM, and storing the ADPCM code in flash memory. Afterward, the ADPCM code will be retrieved from the flash memory, decoded, and the decompressed audio saved back to flash. Both the ADPCM code and the decompressed audio data will be written to the flash memory for analysis of audio quality. The flash memory has a capacity of 8 megabytes, and since the original uncompressed audio file is approximately 938 kilobytes, space will be allocated as follows: the original audio will be stored at the beginning of the flash memory, the ADPCM code will start at page 8192