Speech Compression by using Adaptive Differential Pulse Code Modulation (ADPCM) technique With Microcontroller

Abstract

Compression is a process of reducing an input data (Speech Signal) bit stream into a small bit size with high quality. Analog signal is a continuous signal takes more space to store the data in memory devices with original size (Bit). All sensor data (Analog Data) stored in computer with original size (Bit), but because of compression technique we store the same data in reduced format we same quality. In compression the unwanted data is eliminate. The main purpose of speech compression is to reduce the data bits for transmission of original data from one place to other & store this data that maintaining the quality as same as original signal. In this compression technique the analog to digital conversion (ADC) process played important role, because of analog to digital conversion analog to digital conversion (ADC) we get quantized sample signal. In that sample signal high correlation property is present between the sampled speech signal. The Adaptive Delta Pulse Code Modulation (ADPCM) techniques use the high correlation property of sampled data for compression of signal. This algorithm cannot compress the sampled data as it. It takes the difference between the predicted sample signal and actual sample signal then encode this difference signal which is explained in details below. The Adaptive Delta Pulse Code Modulation (ADPCM) methods have very efficient methods for the compression of signal by reduction of number of bits per sample from original signal with maintaining the quality of signal.
There are so many data compression technique available, but some technique algorithm operation not gives actual quality of signal after compression. That type of technique is called as lossy type algorithm. Because this type lossy algorithm the human ear cannot detect the word. Human voice frequency ranges from 300 Hz to 3400 Hz. Adaptive Delta Pulse Code Modulation (ADPCM) 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 compression & decompression devices.
In this project we used ARM controller which is heart of this project that contains 10-bit channel analog to digital conversion (ADC) pin. Means we get the sample upto 1024. this sample we have going uses for the Adaptive Delta Pulse Code Modulation (ADPCM) algorithm. Also in ARM controller Digital to Analog coversion (DAC) pin has available to check the compressed signal with original signal. Because of ARM controller we reduce the circuitry. Also, we use the Digital signal oscilloscope (DSO) & personal computer to check the behavioural of signal.
Because of compression we save the memory & transmission time with same quality. Also when we want this data we check from stored from memory. In so many government offices, private colleges, laboratory & industry requires to store the original data in computer or in memory devices as it is, so many memory devices are required hence wastage of money is take place. Because of compression by using adaptive differential pulse code modulation technique with microcontroller we achieve the compression of signal with same quality.