運用G.729與G.723.1於多點會議系統之多聲道語音混合方法

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姓名

陳慶彰(Ching-Chang Chen ) 查詢紙本館藏

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電機工程研究所

論文名稱

運用G.729與G.723.1於多點會議系統之多聲道語音混合方法
(A G.729 and G.723.1 Based Multi-Channel Speech Mixing Method for Multi-Point Conferencing System)

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摘要(中)

摘要
音訊混合（audio mixing）是網路音訊會議中不可缺少的機制，其重要性除了提供與真實會議現場一樣的發言環境，讓每位與會者在網路上以全雙工（full-duplex）的方式進行交談，尚可應用在網路連線遊戲及語音聊天室等娛樂用途。
傳統上最佳的語音混合方法是使用完全解碼（full decoding）的方式，在過程上必需進行語音的壓縮及解壓縮處理，造成運算複雜度過高與時間延遲長的缺點。為此，本論文提出一套部份解碼（partial decoding）方式的語音混合方法，利用語音訊號的特性，針對多個待混合的已壓縮音訊訊號，以碼框（frame）為單位，分析代表各音訊所需的音訊參數，選出一組目標音訊參數，作為混合後的音訊輸出。該組目標音訊參數亦符合原壓縮方法之壓縮格式，同時，可混合多個輸入的音訊。可運用在G.729與G.723.1語音壓縮標準上，並有效地降低運算複雜度為完全解碼法的5%至8%，且可得到與完全解碼法相同的混音品質。

摘要(英)

In a multi-point conference, users are offered a substitute for a face-to-face meeting within the economic constraints of the technology available. In this situation, an audio mixing scheme is needed to make the meeting successful. Audio mixing can create a full-duplex conversation environment that users can speak at any moment. Furthermore, it can be used in entertainment applications, such as audio chat rooms and online games.
Full decoding method is an intuitive and traditional audio mixing method, but it requires high computational complexity and long processing time. In this work, we propose a partial decoding method based on CELP coding architecture. This method selects a target frame as the mixed output from all incoming frames. There is no need for any encoding and decoding processes. Partial decoding method can be directly applied to CELP based speech coding, such as G.729 and G.723.1 speech standards. It achieves excellent voice quality as the full decoding method does while it only requires 5% to 8% computation loading.

關鍵字(中)

★ 多點會議系統
★ 完全解碼
★ 部份解碼
★ 音訊混合

關鍵字(英)

★ Audio Mixing
★ Full Decoding
★ Multipoint Conference
★ Partial Decoding

論文目次

內容頁碼
中文摘要 .………...……..……………………………………………. I
Abstract …………………………………………………….………… II
目錄 ……………………………………………………….………… III
附圖索引 ………………………………………………….………. VI
附表索引 ……………………………………………….………. IX
第一章緒論 ………………………..……………….………… 1
1.1多點會議系統簡介 ………………………………………… 1
1.2研究動機與目的 ………………..………………………….. 4
1.3論文架構 ……………….…………………..………………... 7
第二章語音編碼技術…………..…………………………… 9
2.1語音的特性 ………….…….……………..…………… 9
2.2線性預測編碼（Linear Prediction Coding） ….……… 11
2.3基頻估測（Pitch Prediction） ……………………………. 16
2.4 CELP編碼理論 ……………………………………...….…. 19
2.5語音編碼系統的特性 …………………………………….. 24
2.5.1位元率 ……………………………...……………...….…. 24
2.5.2時間延遲 ……………………………………………...…. 26
2.5.3運算複雜度 ……………….…………………………..…. 28
2.5.4語音品質 ………………….…………………………..…. 28
第三章 G.729與G.723.1差異性研究…….……….…… 31
3.1線性預測分析及量化（Linear Prediction Analysis
and Quantization） ………………………..…………… 33
3.2感觀權重濾波器（Perceptual Weighting Filter） ..….…. 38
3.3開迴路基頻分析（Open Loop Pitch Analysis） ……….. 40
3.4適應性碼簿搜尋（Adaptive-codebook Search） ………. 42
3.5固定性碼簿之結構與搜尋 …………………………....…. 47
3.6增益量化 ……………………………………….………...…. 53
3.7附錄介紹 …………………………………………….. 55
第四章多聲道語音混合方法之探討 …………….……59
4.1高位元率語音壓縮的混合方法…………..…………...…. 61
4.1.1線性疊加………………..……………..……………..…… 61
4.1.2時間分割多工…………..……………..……………..…… 64
4.2低位元率語音壓縮的混合方法…………………….……..65
4.2.1完全解碼法……………..……………..……………..…… 65
4.2.2部份解碼法……………..……………..……………..…… 66
4.2.2.1雙聲道語音混合……………….…………..…………68
4.2.2.2多聲道語音混合……………….…………..…………72
4.2.3樹狀結構部份解碼法………………....……………..…… 73
4.3語音混合之品質評量方法……………….…………... 74
4.3.1客觀品質評量………………………....……………..…… 74
4.3.2主觀品質評量………………………....……………..…… 76
第五章模擬實驗與品質評估…………………………….. 77
5.1模擬環境及語音資料…..………..……………………...… 77
5.2低位元率語音壓縮的混合品質 …………………..…..… 79
5.3主觀聽覺測試 ……………………………...…..………...… 95
5.4運算複雜度分析 ………………………………………...… 96
第六章結論 ……………………………………….….……..… 98
參考文獻 ……………………………………………………...… 99

參考文獻

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指導教授

張寶基(Pao-Chi Chang)

審核日期

2001-6-13

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