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The Implementation of Speech Coding System Based on Microcontroller

Author: ZhangChongWu
Tutor: YuanDongFeng; LiuYong
School: Shandong University
Course: Electronics and Communication Engineering
Keywords: Speech Coding SCM ADPCM
CLC: TN912.3
Type: Master's thesis
Year: 2012
Downloads: 277
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Abstract


In recent years, with the rapid development of digital signal processing technology, the speech codec technology has been progressing constantly. Voice information is compressed by encoding to reduce transmission bandwidth and memory space that voice information occupies. In this case, information flow shares less bandwidth in the channel. Compression ratio, Complexity of the algorithm and SNR etcetera are the most basic indicators to measure an algorithm. The in-depth research of various encoding algorithms makes the quality of modern communications quite high level never before.ADPCM further compresses voice information on the basis of pulse code modulation (PCM) adopting advanced adaptive quantization and adaptive prediction techniques. It has the most optimal performance in all waveform encoding algorithms. The complexity of ADPCM algorithm is not very high, so it can be easily implemented relatively. Most of current speech systems are implemented with the special encoding chips or the DSP platforms which have high performance and high system cost at the same time. In some low-end voice communication fields, it is not an economic and optional way. SCM is a cheap, efficient and reliable micro-control device which has been widely used in current electronic field. This paper attempts to achieve the high-performance ADPCM speech encoding system in microcontroller PIC16C72. By this way we can meet the low-speed voice communication need of low-power, low cost, low bandwidth and high-quality in practical engineering.This paper starts with the consideration of the codec status, describing the background of research project and analyzing the purpose and significance of research. Then it describes the development history of speech encoding from rise to today in detail, introduces currently used encoding principles and their characteristics generally, and tells encoding rate, subjective and objective evaluation systems and the encoding delay and so on at the same time. In the last of this part it explores the future development directions of speech encoding. Next it details the principles of ADPCM encoding from the general sample and quantification to ADPCM with the adaptive quantization and the adaptive prediction step by step. Finally, through the design of software and hardware, the author builds the ADPCM voice codec system which takes the microcontroller PIC16C72as a core successfully and also makes an in-depth analysis and performance testing.The low-pass filters used in the voice system are selected based on the simulation results to consider a compromise. In practice, there must be better filters that can be applied to the voice systems. Therefore, the voice codec system based on microchip remains to be further research, and its performance will be better and better.

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CLC: > Industrial Technology > Radio electronics, telecommunications technology > Communicate > Electro-acoustic technology and speech signal processing > Speech Signal Processing
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