基於多重敘述編碼理論之無線通訊系統的品質管理研究

Abstract

無線通訊的服務品質取決於諸多因素，包括封包漏失、延遲時間、背景雜訊、及語音編碼失真。本篇論文旨在探討多重敘述編碼理論之無線通訊系統的品質管理研究，主要採用多重敘述傳輸系統，一方面利用路徑分集以增加傳輸系統的強健性，另一方面可利用不同敘述間的相關性設計其錯誤隱匿機制。有關傳輸位元錯誤的隱匿機制，前人研究基於強健性能與快速實現的整體考量，根據渦旋碼理論而發展疊代訊源通道解碼演算法，關鍵元件包括軟性輸出通道解碼器和軟性位元 訊源解碼器。問題是一般採用的位元層級通道解碼演算法有其限制，不僅無法將相鄰索引之間的相關特性有效整合於訊源事前訊息，與基於索引層級而推導的訊源解碼演算法也存在著相容性的問題。針對這些議題，本論文研究將鎖定索引層級的疊代訊源通道解碼機制。首先開發一個索引層級的BCJR 通道解碼演算法，可有效整合訊源的事前訊息於其軟性輸出的解碼過程。並且進一步配合多重敘述所屬相關訊息的交叉運用，準確估算不同傳輸索引值的後驗機率，並依最小均方 誤差準則求得多重敘述向量量化的最佳解碼輸出。另一個重要的議題則是接收端播放緩衝器的設計。系統設計應整體考量不同關鍵元件的最佳組合，且因應隨時變化的網路傳輸特性作合理調整。首先，我們延伸國際電信聯盟ITU 針對單一路徑傳輸系統所制訂的E-model，進一步開發新的音質評量效能指標，可廣泛應用在多重敘述傳輸的系統規劃。有別於前人研究是將播放緩衝器與前向錯誤控制分開設計，本研究基於音質最佳化的設計理念提出一個適應性整合控制演算法。根 據新的音質評量指標，多重敘述傳輸系統的設計規劃成為一個音質損害最小化問題，依據網路動態彈性調整前向錯誤控制與播放排程，進而達到延遲與封包漏失的最佳平衡點。

Packet loss and network delay are two essential problems to real-time voice communication over mobile IP Networks. The purpose of this dissertation is to develop a multi-stream voice communication system with its quality of service (QoS) control for increased channel robustness. The first part will focus on the error concealment of packet-erasure as well as channel bit errors. The basic strategy is a multiple description scalar quantization (MDSQ) system, in which multiple correlated indexes of the source are assigned and transmitted over channels to take advantage of largely uncorrelated loss and delay characteristics. We propose the use of turbo principle to develop a symbol-based iterative source-channel decoding algorithm for better decoding of multiple descriptions over a noisy channel. We first modify the BCJR algorithm based on sectionalization trellis so that symbol a posteriori probabilities can be derived and used as the extrinsic information to improve the iterative decoding between the source and channel decoders. The residual source redundancies are exploited as a priori informa- tion and a joint source decoding is formulated in the form of a maximum a posteriori estimation problem. We also formulate a recursive implementation for the source de- coder that processes reliability information received on different channels and combines them with inter-description correlation to estimate the transmitted quantizer indexes. Another important issue to address is the playout buffer design which is used at the receiver to smooth out the jitter. As a further step toward perceptual optimization, the error concealing capabilities of multiple description coding can be improved by including an forward error control (FEC) mechanism. We present an objective method for multi-stream voice quality prediction model. Based on the new prediction model, we proposed the use of minimum overall impairment as a perceptually motivated op- timization criterion for joint playout buffer and FEC control. Joint playout and FEC adjustment is then formulated as an optimization problem leading to a better balance between end-to-end delay and packet loss.