G.723.1語音編碼器在ARM晶片之實現

Abstract

本論文主要在研究G.723.1語音編碼器的原理並將其實現在目前被廣泛研究及應用的微處理器ARM上。隨著網路傳輸技術的發達，在無線通訊及3C產品上的應用已經是不可或缺的基本功能，而語音壓縮則是多媒體聲音訊號處理中值得注意的一項技術。G.723.1 語音編碼標準是由ITU-T (International Telecommunication Union - Telecommunication Standardization Sector) 於1996年所制定，具有5.3、6.3 kbit/s兩種壓縮率。它的概念源自於CELP (Code-Excited Linear Prediction) 壓縮法，主要在針對語音及多媒體音訊做壓縮處理，可用於網路電話、視訊會議，並且被採用為MPEG-4規格中的一個標準。 本論文開始於G.723.1語音編碼器的原理介紹，針對主要架構加以闡述及推導，並逐一說明運作方式。接著說明以ARM微處理器實現的過程。在發展的過程中為求達到即時實現的目標，我們改寫C原始程式並修改組合語言，以期能充分發揮ARM的特性。經過各方面的調整後，運算量可大幅降低，配合ARM9微處理器可達到即時處理的效果。

In this thesis, we investigate a speech coding standard, G.723.1, and implement it on a ARM processor. With the invention and progress on the internet, multimedia has become essential in wireless communication and 3C products. Audio and vedio are two aspects of multimedia applications and speech coding is one sort of the former. G.723.1 is a speech coding standard produced by ITU-T (International Telecommunication Union - Telecommunication Standardization Sector) in 1996 for compressing the speech or other audio signal component of multimedia services at a very low bit rate. It is designed on the basis of CELP concept with dual rate: 5.3、6.3 kbit/s and can be widely used in the video-phone, the internet phone phone, and particularly as part of the MPEG-4 standard. At first, coder and decoder principles in G.723.1 will be described and explained in this thesis respectively. Then its implementation will be realized step by step. With an eye to accomplish the real-time purpose, some modifications are performed on either C code or assembly code to bring all advantages of ARM processor into full play. Finally, the aim of real-time implementation is attained with ARM9 processor.