An Investigation on Low Bit-Rate Speech Coding and Its Real-Time Implementation

碩士 === 國立交通大學 === 電子工程系 === 88 === The speech compression technology is an essential element in a wireless multimedia communication system. The goal of this thesis is to investigate the channel robustness properties of the standard speech coding algorithms and their real-time implementati...

Full description

Bibliographic Details
Main Authors: Cheng Han Yan, 楊政翰
Other Authors: Hsueh Ming Hang
Format: Others
Language:zh-TW
Published: 2000
Online Access:http://ndltd.ncl.edu.tw/handle/93003379120127295638
Description
Summary:碩士 === 國立交通大學 === 電子工程系 === 88 === The speech compression technology is an essential element in a wireless multimedia communication system. The goal of this thesis is to investigate the channel robustness properties of the standard speech coding algorithms and their real-time implementations. Essentially, two types of low bit-rate speech coding schemes have been investigated. One is the ITU-T G.723.1 dual rate speech coding, which is designed based on Code-Excited Linear Predictive Coding (CELP). The other is Harmonic Vector eXcitation Coding (HVXC), which is a part of the MPEG-4 specifications. In this thesis, we first evaluate the quality of these two speech coding schemes. Experiments show that the objective quality of the G.723.1 6.3kbit/s speech coding scheme is better than that of the MPEG-4 HVXC 4.0kbit/s speech coding scheme, but the subjective quality of these two speech coding schemes are comparable. Then, we take the noise effects into consideration because the wireless channel is generally very noisy. We compare these two algorithms using two channel error models, additive White Gaussion Noise channel (AWGN) and Markov channel (Gilbert). Our simulation indicates that the objective and subjective speech quality of both schemes degrade rapidly in a highly noisy environment. In order to improve the speech quality, an error concealment technology is proposed for the G.723.1 scheme. The results show that both the objective and subjective quality of speech after this concealment post-processing are significantly improved. Moreover, we implement the G.723.1 speech coding scheme using the Texas Instrument (TI) TMS320C6201 fixed-point digital signal processor (DSP). In order to speed up the DSP implementation, we need to modify the C source programs to take the advantages of the TI DSP features. After proper tuning and optimization on the source programs, the total computation consumption for the encoding and decoding process is about 0.9 million instruction cycles, which is 1.7% of the original computation without tuning.