臺灣博碩士論文加值系統

隨著網路的使用與多媒體技術的快速成長，現今多媒體通訊已被整合於個人行動裝置中，由於低階行動裝置的運算處理速度有限，需要具有低計算複雜度之語音編碼器才能適用於低階行動裝置的硬體平台及整合多媒體資源的服務。對網路或是無線語音通訊器，大量的編碼計算需要消耗更多的功率，需求更高單價的通訊器，減少電池壽命。為了實現語音通訊的即時性和連續性，對於現代通訊系統來說，期望減少語音編碼器的計算複雜度。因此，本論文提出使用語音活動檢測（VAD）演算法，將語音訊號分成活動音框與非活動音框。
在實驗中我們分析非活動語音訊號的特性，發現非活動語音訊號的編碼參數在碼簿結構呈現均勻分佈。因此，當音框為非活動語音訊號時，則不編碼該音框的隨機碼簿激發訊號，而是隨機採用碼簿內的參數代替該音框的隨機碼簿激發訊號，達到減少語音編碼器的編碼計算量。實驗結果證明，整體的客觀語音品質評估平均略微降低0.023，且本文提出的方法相對於原始G.723.1編碼器的計算複雜度，可以降低總計算複雜度大約30％，主觀聽覺上感受不出語音品質下降。

The explosive growth of Internet use and multimedia technology, multimedia communication is integrated into a personal information machine nowadays, and due to the latter’s limited computational capability, the need for a coder with low computational complexity to match different hardware platforms and integrate the services of media sources has arisen. For an Internet or wireless speech communicator, heavy computation uses more power and contributes to higher pricing of the communicator or reduced battery life. In order to achieve the real-time and continuity of speech communication, reduction of computational complexity for the speech coder is desirable for modern communication systems. In this thesis, we use a Voice Activity Detection (VAD) algorithm, which is merely used to classify the speech signal into two types of frames, active frames and inactive frames in our proposed method.
We analyzed the characteristic of the inactive speech signals in our experiments. The experimental results are obvious that the encoding parameters are uniform distributed for the inactive speech subframes. Therefore, if the current frame is an inactive speech frame, then the code excited signal of current frame is not encoded instead of random arrangement the encoding parameters for the codebook structure. The Overall simulation results indicate that the average perceptual evaluation of speech quality score is degraded slightly, by 0.023, and our proposed methods can reduce total computational complexity by about 30% relative to the original G.723.1 encoder computation load with perceptually negligible degradation.

摘 要
ABSTRACT
致 謝
目 錄
表目錄
圖目錄
第1章 簡介
1.1 語音編碼技術背景
1.2 ITU-T制定語音編碼標準
1.3 論文研究目的
1.4 論文大綱
第2章 G.723.1語音編碼器
2.1 CELP編碼架構
2.2 ITU-T G.723.1語音編碼器
2.3 音框處理(Framer)
2.4 高通濾波器(High Pass Filter)
2.5 線性預估編碼分析(LPC Analysis)
2.6 線頻譜對量化(LSP Quantizer)
2.7 共振峰感官加權濾波器(FPWF)
2.8 開迴路基週預估(Pitch Estimator)
2.9 閉迴路適應性基週預估器(Pitch Predictor)
2.10 諧波雜訊濾波器(Harmonic Noise Shaping)
2.11 代數碼激發線性預估(ACELP)
2.12 語音活動檢測(VAD)
2.13 語音品質評估(PESQ)
第3章 非活動語音訊號編碼參數特性之分析
3.1 閉迴路適應性基週預估器編碼參數特性之分析
3.1.1 五階閉迴路適應性基週(pitch lag)特性之分析
3.1.2 五階閉迴路適應性基週增益(gain)特性之分析
3.2 ACELP 碼簿參數特性之分析
3.2.1 ACELP激發脈衝位置與極性之特性分析
3.2.2 ACELP激發脈衝增益(Gain)特性之分析
第4章 計算複雜度與語音品質的評估實驗
4.1 隨機閉迴路基週參數對音質的評估實驗
4.2 隨機ACELP碼激參數對音質的評估實驗
4.3 非活動語音訊號使用隨機編碼參數對音質之評估
第5章 結論
參考文獻
符號彙編
Computational Complexity Reduction of G.723.1 oder Using a voice activity detection Algorithm
Efficient Reduction Computational Complexity For Speech Coding
作者簡介

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[19]ITU-T Rec. G.723.1: Dual Rate Speech Coder for Multimedia Communications Transmitting at 5.3 and 6.3 kbps, March 1996.

[20]ITU-T Rec. H.323: Visual Telephone Systems And Equipment for Local Area Networks Which Provide A Non-Guaranteed Quality of Service, November 1996.

[21]ITU-T Rec. H.324: Terminal for Low Bit Rate Multimedia Communication, March 1996.

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[23]ITU-T Rec. P.862: Perceptual Evaluation of Speech Quality (PESQ): An Objective Method for End-to-End Speech Quality Assessment of Narrow-Band Telephone Networks And Speech Codecs, Feb 2001.

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