隨著網際網路的發展和成熟，各式各樣的網路應用隨之興起，視訊串流服務也是最熱門的項目之一。然而在非可靠的網際網路環境下，難以避免的封包丟失使得視訊資料無法保證被即時或順利地傳送。因此在視訊串流的服務中，系統對於視訊資料傳送的時間延遲，以及網路封包丟失的掌控性特別地要求。採用錯誤更正碼在視訊資料傳送前進行編碼保護，為一種目前主流的解決方案。Digital Fountain codes是無限編碼率的錯誤更正碼，其解碼率以機率形式表現，其中最著名的是由Luby所提出的LT codes。LT codes的編/解碼皆可在線性於視訊資料量的時間內完成，並且實作上容易而簡單。可惜的是，LT codes並無法保證可解碼資料的選擇性，對於不同部分擁有相異重要性的視訊串流資料而言，直接地套用並非是好的解決方案。另外，根據LT codes的理論，唯有當原始資料的數量接近無限多，才能確保擁有最小的編碼冗餘度，導致實際運用在有限數量的視訊串流資料時，編碼冗餘度顯得過高而不盡理想。加上網路環境的不可靠性，一旦LT codes解碼器無法接收到一定數量的編碼資料，將致使解碼效能突然呈現嚴重地衰退。因此，我們試圖透過改良LT codes的編碼結構圖形，以適應重要性差異的視訊串流資料特性，並期待提出一般化的重要性差異視訊資料編碼解決方案，使得在非可靠性網路傳輸下，達到較佳的服務品質；並且即使面對各種異質的客戶端，視訊串流伺服器只需以一種編碼方式，便得以服務所有客戶端的需求，並幫助多數的使用者盡早解完所有的資訊。論文最後的模擬實驗中，顯示了我們的解決方案在重要性差異保護效能上勝過了採用LT codes架構的方案，對視訊串流資料在傳輸保護上取得較佳的優勢。並且面對有限數量的視訊串流資料時，我們的解決方案只需要較小的編碼冗餘度。此外，即使面對嚴峻的傳輸環境，其解碼效能亦能呈現和緩地漸次衰退。

When it comes to multimedia communications, they are considerably sensitive to delay time and packet erasures. Digital Fountain codes are becoming increasingly important thanks to the ability to protect the source data in high probability. The most representative Digital Fountain codes are LT codes. With LT codes, the number of encoding packets can be decided on-the-fly and it is considerably favorable characteristic especially in broadcast channel. It is because different end-users actually suffer from distinct channel loss rates. However, LT encoder simply generates the encoding packets through randomly chosen encoding degree so that it cannot control or manage the service quality according to the distinct service demands of clients. In this thesis, we proposed a namely “service-driven approximate LT codes” that using numerous basic building structures to form the coding graph. The proposed encoder arranges the encoding degree through the basic building structure to generate encoding packets in accordance with various transmission policies and the overall degree distribution approximates to Ideal Soliton distribution. Through our investigations, we found that after transmitting a number of encoded packets in broadcast channel, the clients under low channel loss rates need more high-degree encoding packets to achieve fully-decoded, whereas others need more low-degree packets to recover more source data due to the lack of receiving sufficient encoding packets. Hence, the proposed encoder ensures to transmit low-degree encoding packets first to significantly cut down the average encoding degree then switches to generate high-degree encoding packets if the majority of clients decode a large percentage of source data to always satisfy the demands of the majority first. Furthermore, comparing to LT codes in point-to-point protocol, the proposed codes also introduce lower decoding code overhead in terms of short source data block length, better intermediate performance to achieve graceful quality degradation over a wide range of channel loss rates, and built-in unequal error protection property through appropriately arranging the placements of source data in the coding graph. Last but not least, similar to LT codes, the decoding complexity of the proposed codes is rather low so that it can be applied on real-time decoders, such as mobile phones, which only offer limited computational power and strictly delay-time constrained.

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