本文關(guān)鍵詞： IP網(wǎng)絡(luò)電話 丟包 自適應(yīng)多速率語音編碼 評價模型 仿真　出處：《東北大學(xué)》2014年碩士論文　論文類型：學(xué)位論文

【摘要】：網(wǎng)絡(luò)語音是因特網(wǎng)語音發(fā)展的代表標(biāo)志,它也可稱為因特網(wǎng)語音(VoIP-Voice over Internet Protocol)技術(shù)。VoIP是在IP分組交換網(wǎng)絡(luò)的基礎(chǔ)上,來實現(xiàn)語音通信的傳輸技術(shù),由于價格便宜和功能種類多,所以得到了廣泛地應(yīng)用。盡管VoIP在成本節(jié)省和服務(wù)改進(jìn)方面提供了各種優(yōu)勢,但部分由于質(zhì)量問題,其推廣一直不太順利。事實上,在傳統(tǒng)電話系統(tǒng)和新型的VoIP系統(tǒng)之間存在著若干基本差異,網(wǎng)絡(luò)的流暢與堵塞必然引起語音信號的延時、抖動、丟包、回聲等現(xiàn)象,影響VoIP網(wǎng)絡(luò)通話質(zhì)量的主要原因就是丟包現(xiàn)象。在接收端,經(jīng)過語音解碼后的語音實時回放的質(zhì)量的損失和劣化就是由于語音包的丟失所帶來的。論文通過自適應(yīng)多速率(AMR)語音編碼技術(shù)在丟包處理上的算法,在VoIP系統(tǒng)中的信源信道,通過自適應(yīng)的聯(lián)合編碼來實現(xiàn)丟包問題的解決方法。這種算法就是采用動態(tài)的網(wǎng)絡(luò)狀況,在信道編碼上選取在Reed-Solomon碼的前向糾錯體制,運用E-Model評價模型對語音質(zhì)量進(jìn)行評估,然后自主選擇一種能夠達(dá)到語音質(zhì)量最好的信源和信道編碼速率。本文設(shè)計了一種自適應(yīng)算法,適用于VoIP系統(tǒng)的信源信道速率,采用了AMR語音編碼技術(shù)和基于RS碼的前向糾錯信道編碼。這種算法通過E-Model評價模型從丟包和延時兩點來判斷語音的質(zhì)量,依據(jù)當(dāng)前變化的網(wǎng)絡(luò)狀況自適應(yīng)地選擇出語音質(zhì)量最優(yōu)的信源與信道編碼方案。對于這種算法,以VisualC++6.0平臺進(jìn)行算法的分析,分別在不同的網(wǎng)絡(luò)狀態(tài)下,求解出使得語音質(zhì)量最優(yōu)時的信源信道速率以及前向糾錯編碼方式,如果與假定的結(jié)果相符,那么就證明了它的有效性。實驗在NS2仿真環(huán)境下進(jìn)行,模擬網(wǎng)絡(luò)環(huán)境的變化,在VoIP系統(tǒng)中,結(jié)合信源、信道速率自適應(yīng)算法,并與VoIP系統(tǒng)中固定速率G.729的編碼器、VoIP系統(tǒng)中信源速率自適應(yīng)AMR的編碼器相比較。試驗表明,論文設(shè)計的這種算法對于丟包和延時兩個問題的影響都不同程度的下降,尤其是在網(wǎng)絡(luò)擁塞的時候它的效果更加明顯。
[Abstract]:Network voice is the representative symbol of Internet voice development. It can also be called VoIP-Voice over Internet Protocol. VoIP is based on IP packet switching network. The transmission technology to realize voice communication is widely used because of its low price and wide variety of functions, although VoIP provides various advantages in terms of cost saving and service improvement. In fact, there are some basic differences between the traditional telephone system and the new VoIP system. The fluency and blockage of the network will inevitably cause the delay, jitter, packet loss, echo and other phenomena of the voice signal. The main reason that affects the quality of the VoIP network is the loss of the packet. The loss and deterioration of the quality of real-time playback after speech decoding is due to the loss of speech packets. This paper uses adaptive multi-rate AMR-based speech coding technology in packet loss processing algorithm. In the source and channel of VoIP system, the solution of packet loss problem is realized by adaptive joint coding. This algorithm adopts dynamic network condition. In the channel coding, the forward error correction scheme of Reed-Solomon code is selected, and the evaluation model of E-Model is used to evaluate the speech quality. Then we choose a source and channel coding rate which can achieve the best speech quality. In this paper, we design an adaptive algorithm, which is suitable for VoIP system. AMR speech coding technology and forward error correction channel coding based on RS code are adopted. This algorithm judges the quality of speech by E-Model evaluation model from packet loss and delay two points. According to the current changing network conditions, the source and channel coding scheme with the best speech quality is adaptively selected. For this algorithm, the algorithm is analyzed on the platform of VisualC 6.0. In different network states, the source channel rate and forward error correction coding method are obtained when the speech quality is optimal, if the results are consistent with the assumed results. The experiment is carried out in the NS2 simulation environment to simulate the change of the network environment. In the VoIP system, combining the source and channel rate adaptive algorithm. It is compared with the source rate adaptive AMR encoder in the fixed rate G. 729 encoder of VoIP system. The effects of this algorithm on packet loss and delay are reduced in different degrees, especially when the network is congested.
【學(xué)位授予單位】：東北大學(xué)
【學(xué)位級別】：碩士
【學(xué)位授予年份】：2014
【分類號】：TN912.3

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