基於空間特徵之可調音訊編碼架構; Spatial characteristic based scalable audio coding structure

NCUIR > College of Electrical Engineering & Computer Science > Graduate Institute of Communication Engineering > Electronic Thesis & Dissertation > Item 987654321/8312

Please use this identifier to cite or link to this item: http://ir.lib.ncu.edu.tw/handle/987654321/8312

Title:	基於空間特徵之可調音訊編碼架構;Spatial characteristic based scalable audio coding structure
Authors:	李宗奇;Tsung-Chi Lee
Contributors:	通訊工程研究所
Keywords:	可調音訊編碼;空間音訊編碼;spatial audio coding;scalable audio coding
Date:	2009-07-08
Issue Date:	2009-09-22 11:22:56 (UTC+8)
Publisher:	國立中央大學圖書館
Abstract:	MPEG Layer-3(MP3) 以及MPEG-4 Advance Audio Coding(AAC)皆具有非常高的編碼效率，主要因素為使用了音響心理學模型(Psychoacoustic Model) 的分析，去除人類聽覺上被遮蔽的頻率成份；然而，音響心理學模型是針對單聲道音訊的分析模型，並沒有考慮多聲道音訊中，聲道之間的相關性，因此，編碼多聲道音訊時，位元率將會隨著聲道的增加而遞增。近年來被大量研究的空間音訊壓縮編碼 (Spatial Audio Coding) 技術，利用人耳在空間中聽覺的特性，抽取對音場感受的特徵參數，在接收端即可以較少的聲道配合特徵參數重建音場。基於目前正在推廣的網際網路電視技術，在網路上傳輸多聲道音訊勢在必行。本論文利用空間音訊壓縮編碼的演算法，架構一個分層可調的音訊編碼，隨著收到更多的增強層，音訊聲道的數量以及品質就會隨之增加，可針對網路的通道狀況進行調整，甚至在必要時調整音訊與視訊的頻寬，給予不同的品質，以符合使用者需求。另外，考慮到多聲道音訊中，若某個聲道出現與其他聲道不相關的音訊內容，將會影響音場重建後的音訊品質，故本論文設計一演算法，解決此問題，使客觀品質量測的分數上升約1分。 MPEG Layer-3 (MP3) and MPEG-4 Advance Audio Coding (AAC) exhibit high coding efficiency by utilizing the psychoacoustic model to remove the masked frequency components. However, the psychoacoustic model aims at the analysis of single channel audio signals without considering the correlation between audio channels. As a result, adding more audio channels to encode will result in an approximately linear increase of the total required transmission bit-rate. The Spatial Audio Coding (SAC) technology exploits human perceptual capability to locate sound in space. It captures and encodes the spatial characteristic parameters at the encoder. At the decoder, the sound field can be reconstructed from fewer audio channels with spatial parameters. In this work, we propose a scalable audio coding scheme, which is based on spatial audio coding techniques including parametric stereo and MPEG surround, to transmit multi-channel audio through networks. More audio channels and better quality can be obtained with more enhancement layers received. We also observe that when uncorrelated signals, such as dialogs, exist in multi-channel signals the reconstructed audio suffers from serious interference. In this case, we execute inter-channel interference processing to encode the uncorrelated part individually. The experimental results show excellent subjective as well as objective quality improvement.
Appears in Collections:	[Graduate Institute of Communication Engineering] Electronic Thesis & Dissertation

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