隨著網際網路的快速發展，在封包網路上傳送聲音資料受到很多的注意。在不可靠的網路上傳送即時資料時，封包遺失是無可避免，而延遲的封包更是增加了這封包的遺失率。這些遺失及延遲的封包會使得聲音播放的品質下降。而且若是使用低位元率的聲音編碼器來編碼，這個下降品質會影響更大。近年來網路也越來越異質，所以可延展的編碼在網路環境中是很重要的。在這裡，我們為低位元率聲音編碼器提出了一個基於多重描述編碼的方法，它可以使得編碼器可延展以及可以隱藏遺失的封包。我們將剩餘訊號分為高頻及低頻兩部分，然後分別使用不同的方法去產生高頻及低頻的增加層。基於取樣的多重描述編碼跟基於線譜對的多重描述編碼都會被我們所用來隱藏遺失的封包。基於取樣的多重描述編碼將會被使用在高頻的增加層，而基於線譜對的多重描述編碼則會被用在核心編碼層跟低頻的增加層。我們實作我們的方法到G.723的標準上。結果顯示出我們提出的方法在不同的網路環境下，都可以提供可階層的功能及好的播放品質。

Voice transmissions over packet networks have attracted attention recently with the quick growth of the Internet. In real-time data transmission over unreliable networks, packet losses are inevitable and the delayed packets further increase the packet loss rate. The lost and delayed packets degrade the playback quality. The degradation of the quality is more perceptible if a low bit-rate speech coder is adopted. Networks are more and more heterogeneous recently. Therefore, the scalable coding is important in network environments. In this thesis, we proposed a robust and scalable scheme based on the MDC schemes for low bit-rate speech coders. We use a split-band approach that decomposes the residual signal into two frequency bands, low-band and high-band. The two bands use different methods to generate low-band and high-band enhancement layers. Both of the sample-based multiple description coding (MDC) scheme and the LSP-based MDC scheme are used for concealing the lost packets. We adopt the sample-based MDC scheme for the high-band enhancement layer and the LSP-based MDC scheme for the core and low-band enhancement layers. We have implemented the proposed scheme on ITU-T G.723.1 standard. The experimental results show that the proposed scheme can provide good quality and scalable capability under different conditions of bit rates and packet loss rates.

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