Abstract
Adaptive Delta Pulse Code Modulation is a well known encoding scheme used for speech processing. This project focuses on the simplification in the technique so that the hardware complexity can be reduced for the portable speech recording devices. The reduce hardware complexity also provides added advantages like low power consumption and portability. The simulation of the suggested simplification in the algorithm is done using the MATLAB tool. This project focuses on .wav format audio file and encoding and decoding using ADPCM technique is done. This project provides practical comparison by comparing the actual audio file with decoded audio file and their respective graphical representations. The hardware implementation of speech recording and playback is done using Atmel STK500 kit and ATMEGA32 microcontroller. The ADC and PWM schemes are used to interface analog signals to the core digital hardware. To conclude, this project tries to provide insight of algorithm and hardware that can be used for the portable speech recording playback devices.